[Freeswitch-svn] [commit] r7544 - in freeswitch/trunk/libs/voipcodecs: . config debian doc etsitests itutests localtests src src/msvc src/msvc/sys src/voipcodecs tests

7 Feb 2008

Author: mikej
Date: Thu Feb  7 17:00:10 2008
New Revision: 7544

Added:
   freeswitch/trunk/libs/voipcodecs/
   freeswitch/trunk/libs/voipcodecs/AUTHORS
   freeswitch/trunk/libs/voipcodecs/COPYING
   freeswitch/trunk/libs/voipcodecs/ChangeLog
   freeswitch/trunk/libs/voipcodecs/INSTALL
   freeswitch/trunk/libs/voipcodecs/Makefile.am
   freeswitch/trunk/libs/voipcodecs/NEWS
   freeswitch/trunk/libs/voipcodecs/README
   freeswitch/trunk/libs/voipcodecs/config/
   freeswitch/trunk/libs/voipcodecs/config-h.in
   freeswitch/trunk/libs/voipcodecs/configure.ac
   freeswitch/trunk/libs/voipcodecs/debian/
   freeswitch/trunk/libs/voipcodecs/debian/changelog
   freeswitch/trunk/libs/voipcodecs/debian/compat
   freeswitch/trunk/libs/voipcodecs/debian/control
   freeswitch/trunk/libs/voipcodecs/debian/copyright
   freeswitch/trunk/libs/voipcodecs/debian/libvoipcodecs-dev.install
   freeswitch/trunk/libs/voipcodecs/debian/libvoipcodecs-doc.install
   freeswitch/trunk/libs/voipcodecs/debian/libvoipcodecs.install
   freeswitch/trunk/libs/voipcodecs/debian/rules   (contents, props changed)
   freeswitch/trunk/libs/voipcodecs/debian/watch
   freeswitch/trunk/libs/voipcodecs/doc/
   freeswitch/trunk/libs/voipcodecs/doc/Makefile.am
   freeswitch/trunk/libs/voipcodecs/doc/css.css
   freeswitch/trunk/libs/voipcodecs/doc/libvoipcodecs-doxygen
   freeswitch/trunk/libs/voipcodecs/doc/wrapper.xsl
   freeswitch/trunk/libs/voipcodecs/etsitests/
   freeswitch/trunk/libs/voipcodecs/etsitests/Makefile.am
   freeswitch/trunk/libs/voipcodecs/itutests/
   freeswitch/trunk/libs/voipcodecs/itutests/Makefile.am
   freeswitch/trunk/libs/voipcodecs/libvoipcodecs.vcproj
   freeswitch/trunk/libs/voipcodecs/localtests/
   freeswitch/trunk/libs/voipcodecs/localtests/Makefile.am
   freeswitch/trunk/libs/voipcodecs/localtests/dam9.wav   (contents, props changed)
   freeswitch/trunk/libs/voipcodecs/localtests/dam9_lpc55.wav   (contents, props changed)
   freeswitch/trunk/libs/voipcodecs/localtests/short_nb_voice.wav   (contents, props changed)
   freeswitch/trunk/libs/voipcodecs/localtests/short_wb_voice.wav   (contents, props changed)
   freeswitch/trunk/libs/voipcodecs/src/
   freeswitch/trunk/libs/voipcodecs/src/Makefile.am
   freeswitch/trunk/libs/voipcodecs/src/bitstream.c
   freeswitch/trunk/libs/voipcodecs/src/float_fudge.h
   freeswitch/trunk/libs/voipcodecs/src/g711.c
   freeswitch/trunk/libs/voipcodecs/src/g722_decode.c
   freeswitch/trunk/libs/voipcodecs/src/g722_encode.c
   freeswitch/trunk/libs/voipcodecs/src/g726.c
   freeswitch/trunk/libs/voipcodecs/src/gsm0610_decode.c
   freeswitch/trunk/libs/voipcodecs/src/gsm0610_encode.c
   freeswitch/trunk/libs/voipcodecs/src/gsm0610_local.h
   freeswitch/trunk/libs/voipcodecs/src/gsm0610_long_term.c
   freeswitch/trunk/libs/voipcodecs/src/gsm0610_lpc.c
   freeswitch/trunk/libs/voipcodecs/src/gsm0610_preprocess.c
   freeswitch/trunk/libs/voipcodecs/src/gsm0610_rpe.c
   freeswitch/trunk/libs/voipcodecs/src/gsm0610_short_term.c
   freeswitch/trunk/libs/voipcodecs/src/ima_adpcm.c
   freeswitch/trunk/libs/voipcodecs/src/libvoipcodecs.dsp
   freeswitch/trunk/libs/voipcodecs/src/libvoipcodecs.sln
   freeswitch/trunk/libs/voipcodecs/src/lpc10_analyse.c
   freeswitch/trunk/libs/voipcodecs/src/lpc10_decode.c
   freeswitch/trunk/libs/voipcodecs/src/lpc10_encdecs.h
   freeswitch/trunk/libs/voipcodecs/src/lpc10_encode.c
   freeswitch/trunk/libs/voipcodecs/src/lpc10_placev.c
   freeswitch/trunk/libs/voipcodecs/src/lpc10_voicing.c
   freeswitch/trunk/libs/voipcodecs/src/msvc/
   freeswitch/trunk/libs/voipcodecs/src/msvc/gettimeofday.c
   freeswitch/trunk/libs/voipcodecs/src/msvc/inttypes.h
   freeswitch/trunk/libs/voipcodecs/src/msvc/msvcproj.foot
   freeswitch/trunk/libs/voipcodecs/src/msvc/msvcproj.head
   freeswitch/trunk/libs/voipcodecs/src/msvc/sys/
   freeswitch/trunk/libs/voipcodecs/src/msvc/sys/time.h
   freeswitch/trunk/libs/voipcodecs/src/msvc/tgmath.h
   freeswitch/trunk/libs/voipcodecs/src/msvc/unistd.h
   freeswitch/trunk/libs/voipcodecs/src/msvc/vc8proj.foot
   freeswitch/trunk/libs/voipcodecs/src/msvc/vc8proj.head
   freeswitch/trunk/libs/voipcodecs/src/msvc/voipcodecs.def
   freeswitch/trunk/libs/voipcodecs/src/oki_adpcm.c
   freeswitch/trunk/libs/voipcodecs/src/vector_int.c
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs.h.in
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/bit_operations.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/bitstream.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/dc_restore.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/g711.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/g722.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/g726.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/gsm0610.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/ima_adpcm.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/lpc10.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/oki_adpcm.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/telephony.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/vector_int.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/version.h
   freeswitch/trunk/libs/voipcodecs/src/voipcodecs/version.h.in
   freeswitch/trunk/libs/voipcodecs/tests/
   freeswitch/trunk/libs/voipcodecs/tests/Makefile.am
   freeswitch/trunk/libs/voipcodecs/tests/g711_tests.c
   freeswitch/trunk/libs/voipcodecs/tests/g722_tests.c
   freeswitch/trunk/libs/voipcodecs/tests/g726_tests.c
   freeswitch/trunk/libs/voipcodecs/tests/gsm0610_tests.c
   freeswitch/trunk/libs/voipcodecs/tests/ima_adpcm_tests.c
   freeswitch/trunk/libs/voipcodecs/tests/lpc10_tests.c
   freeswitch/trunk/libs/voipcodecs/tests/oki_adpcm_tests.c
   freeswitch/trunk/libs/voipcodecs/tests/regression_tests.sh   (contents, props changed)
   freeswitch/trunk/libs/voipcodecs/voipcodecs.spec.in
   freeswitch/trunk/libs/voipcodecs/wrapper.xsl

Log:
inital checkin of voipcodecs library version 0.0.1 from http://www.soft-switch.org/downloads/voipcodecs/ thank you Steve

Added: freeswitch/trunk/libs/voipcodecs/AUTHORS
==============================================================================

--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/AUTHORS	Thu Feb  7 17:00:10 2008
@@ -0,0 +1 @@
+Steve Underwood <steveu@coppice.org>

Added: freeswitch/trunk/libs/voipcodecs/COPYING
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/COPYING	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,340 @@
+		    GNU GENERAL PUBLIC LICENSE
+		       Version 2, June 1991
+
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Added: freeswitch/trunk/libs/voipcodecs/ChangeLog
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/ChangeLog	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,3 @@
+08.02.08 - 0.0.1 - Steve Underwood <steveu@coppice.org>
+    - The first version.
+

Added: freeswitch/trunk/libs/voipcodecs/INSTALL
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/INSTALL	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,182 @@
+Basic Installation
+==================
+
+   These are generic installation instructions.
+
+   The `configure' shell script attempts to guess correct values for
+various system-dependent variables used during compilation.  It uses
+those values to create a `Makefile' in each directory of the package.
+It may also create one or more `.h' files containing system-dependent
+definitions.  Finally, it creates a shell script `config.status' that
+you can run in the future to recreate the current configuration, a file
+`config.cache' that saves the results of its tests to speed up
+reconfiguring, and a file `config.log' containing compiler output
+(useful mainly for debugging `configure').
+
+   If you need to do unusual things to compile the package, please try
+to figure out how `configure' could check whether to do them, and mail
+diffs or instructions to the address given in the `README' so they can
+be considered for the next release.  If at some point `config.cache'
+contains results you don't want to keep, you may remove or edit it.
+
+   The file `configure.ac' is used to create `configure' by a program
+called `autoconf'.  You only need `configure.ac' if you want to change
+it or regenerate `configure' using a newer version of `autoconf'.
+
+The simplest way to compile this package is:
+
+  1. `cd' to the directory containing the package's source code and type
+     `./configure' to configure the package for your system.  If you're
+     using `csh' on an old version of System V, you might need to type
+     `sh ./configure' instead to prevent `csh' from trying to execute
+     `configure' itself.
+
+     Running `configure' takes awhile.  While running, it prints some
+     messages telling which features it is checking for.
+
+  2. Type `make' to compile the package.
+
+  3. Optionally, type `make check' to run any self-tests that come with
+     the package.
+
+  4. Type `make install' to install the programs and any data files and
+     documentation.
+
+  5. You can remove the program binaries and object files from the
+     source code directory by typing `make clean'.  To also remove the
+     files that `configure' created (so you can compile the package for
+     a different kind of computer), type `make distclean'.  There is
+     also a `make maintainer-clean' target, but that is intended mainly
+     for the package's developers.  If you use it, you may have to get
+     all sorts of other programs in order to regenerate files that came
+     with the distribution.
+
+Compilers and Options
+=====================
+
+   Some systems require unusual options for compilation or linking that
+the `configure' script does not know about.  You can give `configure'
+initial values for variables by setting them in the environment.  Using
+a Bourne-compatible shell, you can do that on the command line like
+this:
+     CC=c89 CFLAGS=-O2 LIBS=-lposix ./configure
+
+Or on systems that have the `env' program, you can do it like this:
+     env CPPFLAGS=-I/usr/local/include LDFLAGS=-s ./configure
+
+Compiling For Multiple Architectures
+====================================
+
+   You can compile the package for more than one kind of computer at the
+same time, by placing the object files for each architecture in their
+own directory.  To do this, you must use a version of `make' that
+supports the `VPATH' variable, such as GNU `make'.  `cd' to the
+directory where you want the object files and executables to go and run
+the `configure' script.  `configure' automatically checks for the
+source code in the directory that `configure' is in and in `..'.
+
+   If you have to use a `make' that does not supports the `VPATH'
+variable, you have to compile the package for one architecture at a time
+in the source code directory.  After you have installed the package for
+one architecture, use `make distclean' before reconfiguring for another
+architecture.
+
+Installation Names
+==================
+
+   By default, `make install' will install the package's files in
+`/usr/local/bin', `/usr/local/man', etc.  You can specify an
+installation prefix other than `/usr/local' by giving `configure' the
+option `--prefix=PATH'.
+
+   You can specify separate installation prefixes for
+architecture-specific files and architecture-independent files.  If you
+give `configure' the option `--exec-prefix=PATH', the package will use
+PATH as the prefix for installing programs and libraries.
+Documentation and other data files will still use the regular prefix.
+
+   In addition, if you use an unusual directory layout you can give
+options like `--bindir=PATH' to specify different values for particular
+kinds of files.  Run `configure --help' for a list of the directories
+you can set and what kinds of files go in them.
+
+   If the package supports it, you can cause programs to be installed
+with an extra prefix or suffix on their names by giving `configure' the
+option `--program-prefix=PREFIX' or `--program-suffix=SUFFIX'.
+
+Optional Features
+=================
+
+   Some packages pay attention to `--enable-FEATURE' options to
+`configure', where FEATURE indicates an optional part of the package.
+They may also pay attention to `--with-PACKAGE' options, where PACKAGE
+is something like `gnu-as' or `x' (for the X Window System).  The
+`README' should mention any `--enable-' and `--with-' options that the
+package recognizes.
+
+   For packages that use the X Window System, `configure' can usually
+find the X include and library files automatically, but if it doesn't,
+you can use the `configure' options `--x-includes=DIR' and
+`--x-libraries=DIR' to specify their locations.
+
+Specifying the System Type
+==========================
+
+   There may be some features `configure' can not figure out
+automatically, but needs to determine by the type of host the package
+will run on.  Usually `configure' can figure that out, but if it prints
+a message saying it can not guess the host type, give it the
+`--host=TYPE' option.  TYPE can either be a short name for the system
+type, such as `sun4', or a canonical name with three fields:
+     CPU-COMPANY-SYSTEM
+
+See the file `config.sub' for the possible values of each field.  If
+`config.sub' isn't included in this package, then this package doesn't
+need to know the host type.
+
+   If you are building compiler tools for cross-compiling, you can also
+use the `--target=TYPE' option to select the type of system they will
+produce code for and the `--build=TYPE' option to select the type of
+system on which you are compiling the package.
+
+Sharing Defaults
+================
+
+   If you want to set default values for `configure' scripts to share,
+you can create a site shell script called `config.site' that gives
+default values for variables like `CC', `cache_file', and `prefix'.
+`configure' looks for `PREFIX/share/config.site' if it exists, then
+`PREFIX/etc/config.site' if it exists.  Or, you can set the
+`CONFIG_SITE' environment variable to the location of the site script.
+A warning: not all `configure' scripts look for a site script.
+
+Operation Controls
+==================
+
+   `configure' recognizes the following options to control how it
+operates.
+
+`--cache-file=FILE'
+     Use and save the results of the tests in FILE instead of
+     `./config.cache'.  Set FILE to `/dev/null' to disable caching, for
+     debugging `configure'.
+
+`--help'
+     Print a summary of the options to `configure', and exit.
+
+`--quiet'
+`--silent'
+`-q'
+     Do not print messages saying which checks are being made.  To
+     suppress all normal output, redirect it to `/dev/null' (any error
+     messages will still be shown).
+
+`--srcdir=DIR'
+     Look for the package's source code in directory DIR.  Usually
+     `configure' can determine that directory automatically.
+
+`--version'
+     Print the version of Autoconf used to generate the `configure'
+     script, and exit.
+
+`configure' also accepts some other, not widely useful, options.

Added: freeswitch/trunk/libs/voipcodecs/Makefile.am
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/Makefile.am	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,93 @@
+##
+## VoIPcodecs - a series of DSP components for telephony
+##
+## Makefile.am -- Process this file with automake to produce Makefile.in
+##
+## This program is free software; you can redistribute it and/or modify
+## it under the terms of the GNU General Public License version 2, as
+## published by the Free Software Foundation.
+##
+## This program is distributed in the hope that it will be useful,
+## but WITHOUT ANY WARRANTY; without even the implied warranty of
+## MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+## GNU General Public License for more details.
+##
+## You should have received a copy of the GNU General Public License
+## along with this program; if not, write to the Free Software
+## Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+##
+## $Id: Makefile.am,v 1.29 2007/09/23 12:45:15 steveu Exp $
+
+AM_CFLAGS = $(COMP_VENDOR_CFLAGS)
+
+noinst_SCRIPTS = voipcodecs.spec
+
+MAINTAINERCLEANFILES = Makefile.in
+
+EXTRA_DIST = voipcodecs.spec \
+             wrapper.xsl \
+             libvoipcodecs.vcproj \
+             doc/libvoipcodecs-doxygen \
+             src/float_fudge.h \
+             src/voipcodecs/version.h.in \
+             src/libvoipcodecs.dsp \
+             src/libvoipcodecs.sln \
+             src/msvc/gettimeofday.c \
+             src/msvc/inttypes.h \
+             src/msvc/tgmath.h \
+             src/msvc/unistd.h \
+             src/msvc/sys/time.h \
+             src/msvc/voipcodecs.def \
+             src/msvc/msvcproj.head \
+             src/msvc/msvcproj.foot \
+             src/msvc/vc8proj.head \
+             src/msvc/vc8proj.foot \
+             debian/changelog \
+             debian/compat \
+             debian/control \
+             debian/copyright \
+             debian/libvoipcodecs.install \
+             debian/libvoipcodecs-dev.install \
+             debian/libvoipcodecs-doc.install \
+             debian/rules \
+             debian/watch
+
+if COND_DOC
+    MAYBE_DOC=doc
+endif
+if COND_TESTS
+    MAYBE_TESTS=tests
+endif
+if COND_ITUTESTS
+    MAYBE_ITUTESTS=itutests
+endif
+SUBDIRS = src $(MAYBE_DOC) $(MAYBE_TESTS) $(MAYBE_ITUTESTS)
+
+DIST_SUBDIRS = src doc tests localtests etsitests itutests
+
+faq: faq.xml
+	cd faq ; xsltproc ../wrapper.xsl ../faq.xml
+
+rpm: rpm-build
+
+rpm-build:
+	$(MAKE) -$(MAKEFLAGS) bump.rpm.release
+	$(MAKE) -$(MAKEFLAGS) dist
+	rm -rf rpm/BUILD/*
+	rm -f rpm/RPMS/*/*
+	rm -f rpm/SOURCES/*
+	rm -f rpm/SPECS/*
+	rm -f rpm/SRPMS/*
+	rpm -ta --sign @PACKAGE@-@VERSION@.tar.gz
+
+bump.rpm.release: voipcodecs.spec
+	VERSION="x"; \
+	test -f $(srcdir)/rpm.release && . $(srcdir)/rpm.release; \
+	NEXT_RELEASE=0; \
+	test "$$VERSION" = "@VERSION@" && NEXT_RELEASE="$$RELEASE"; \
+	RELEASE=`expr $$NEXT_RELEASE + 1`; \
+	echo "VERSION=@VERSION@" >$(srcdir)/rpm.release; \
+	echo "RELEASE=$$RELEASE" >>$(srcdir)/rpm.release; \
+	sed 's/^Release: .*/Release: '$$RELEASE'/' \
+		<voipcodecs.spec >voipcodecs.spec.new; \
+	mv voipcodecs.spec.new voipcodecs.spec

Added: freeswitch/trunk/libs/voipcodecs/NEWS
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/NEWS	Thu Feb  7 17:00:10 2008
@@ -0,0 +1 @@
+No news is good news!
\ No newline at end of file

Added: freeswitch/trunk/libs/voipcodecs/README
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/README	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,4 @@
+VoIPcodecs 0.0.1 - A set of commonly used, unencumbered, codecs for VoIP
+------------------------------------------------------------------------
+
+Steve Underwood <steveu@coppice.org>

Added: freeswitch/trunk/libs/voipcodecs/config-h.in
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/config-h.in	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,235 @@
+/* config-h.in.  Generated from configure.ac by autoheader.  */
+
+/* Define to 1 if you have the <audiofile.h> header file. */
+#undef HAVE_AUDIOFILE_H
+
+/* Define to 1 if you have the <dlfcn.h> header file. */
+#undef HAVE_DLFCN_H
+
+/* Define to 1 if you don't have `vprintf' but do have `_doprnt.' */
+#undef HAVE_DOPRNT
+
+/* Define to 1 if you have the <fcntl.h> header file. */
+#undef HAVE_FCNTL_H
+
+/* Define to 1 if you have the <fftw3.h> header file. */
+#undef HAVE_FFTW3_H
+
+/* Define to 1 if you have the <fftw.h> header file. */
+#undef HAVE_FFTW_H
+
+/* Define to 1 if you have the <float.h> header file. */
+#undef HAVE_FLOAT_H
+
+/* Define to 1 if you have the <FL/Fl_Audio_Meter.H> header file. */
+#undef HAVE_FL_FL_AUDIO_METER_H
+
+/* Define to 1 if you have the <FL/Fl_Cartesian.H> header file. */
+#undef HAVE_FL_FL_CARTESIAN_H
+
+/* Define to 1 if you have the <FL/fl_draw.H> header file. */
+#undef HAVE_FL_FL_DRAW_H
+
+/* Define to 1 if you have the <FL/Fl.H> header file. */
+#undef HAVE_FL_FL_H
+
+/* Define to 1 if you have the <FL/Fl_Light_Button.H> header file. */
+#undef HAVE_FL_FL_LIGHT_BUTTON_H
+
+/* Define to 1 if you have the <FL/Fl_Overlay_Window.H> header file. */
+#undef HAVE_FL_FL_OVERLAY_WINDOW_H
+
+/* Define to 1 if you have the `gettimeofday' function. */
+#undef HAVE_GETTIMEOFDAY
+
+/* Define to 1 if you have the <inttypes.h> header file. */
+#undef HAVE_INTTYPES_H
+
+/* Define to 1 if you have the `m' library (-lm). */
+#undef HAVE_LIBM
+
+/* Define to 1 if you have the `tiff' library (-ltiff). */
+#undef HAVE_LIBTIFF
+
+/* Define to 1 if you have the 'libxml2' library (-lxml2). */
+#undef HAVE_LIBXML2
+
+/* Define to 1 if you have the <libxml/parser.h> header file. */
+#undef HAVE_LIBXML_PARSER_H
+
+/* Define to 1 if you have the <libxml/xinclude.h> header file. */
+#undef HAVE_LIBXML_XINCLUDE_H
+
+/* Define to 1 if you have the <libxml/xmlmemory.h> header file. */
+#undef HAVE_LIBXML_XMLMEMORY_H
+
+/* Define to 1 if the system has the type `long double'. */
+#undef HAVE_LONG_DOUBLE
+
+/* Define to 1 if the system has the type `long long'. */
+#undef HAVE_LONG_LONG
+
+/* Define to 1 if your system has a GNU libc compatible `malloc' function, and
+   to 0 otherwise. */
+#undef HAVE_MALLOC
+
+/* Define to 1 if you have the <malloc.h> header file. */
+#undef HAVE_MALLOC_H
+
+/* Define to 1 if you have the <math.h> header file. */
+#undef HAVE_MATH_H
+
+/* Define to 1 if you have the `memmove' function. */
+#undef HAVE_MEMMOVE
+
+/* Define to 1 if you have the <memory.h> header file. */
+#undef HAVE_MEMORY_H
+
+/* Define to 1 if you have the `memset' function. */
+#undef HAVE_MEMSET
+
+/* Define to 1 if you have the <pthread.h> header file. */
+#undef HAVE_PTHREAD_H
+
+/* Define to 1 if your system has a GNU libc compatible `realloc' function,
+   and to 0 otherwise. */
+#undef HAVE_REALLOC
+
+/* Define to 1 if you have the `select' function. */
+#undef HAVE_SELECT
+
+/* Define to 1 if you have the <socket.h> header file. */
+#undef HAVE_SOCKET_H
+
+/* Define to 1 if you have the <stdint.h> header file. */
+#undef HAVE_STDINT_H
+
+/* Define to 1 if you have the <stdlib.h> header file. */
+#undef HAVE_STDLIB_H
+
+/* Define to 1 if you have the `strcasecmp' function. */
+#undef HAVE_STRCASECMP
+
+/* Define to 1 if you have the `strchr' function. */
+#undef HAVE_STRCHR
+
+/* Define to 1 if you have the `strdup' function. */
+#undef HAVE_STRDUP
+
+/* Define to 1 if you have the `strerror' function. */
+#undef HAVE_STRERROR
+
+/* Define to 1 if you have the <strings.h> header file. */
+#undef HAVE_STRINGS_H
+
+/* Define to 1 if you have the <string.h> header file. */
+#undef HAVE_STRING_H
+
+/* Define to 1 if you have the `strstr' function. */
+#undef HAVE_STRSTR
+
+/* Define to 1 if you have the `strtol' function. */
+#undef HAVE_STRTOL
+
+/* Define to 1 if you have the <sys/fcntl.h> header file. */
+#undef HAVE_SYS_FCNTL_H
+
+/* Define to 1 if you have the <sys/ioctl.h> header file. */
+#undef HAVE_SYS_IOCTL_H
+
+/* Define to 1 if you have the <sys/select.h> header file. */
+#undef HAVE_SYS_SELECT_H
+
+/* Define to 1 if you have the <sys/socket.h> header file. */
+#undef HAVE_SYS_SOCKET_H
+
+/* Define to 1 if you have the <sys/stat.h> header file. */
+#undef HAVE_SYS_STAT_H
+
+/* Define to 1 if you have the <sys/time.h> header file. */
+#undef HAVE_SYS_TIME_H
+
+/* Define to 1 if you have the <sys/types.h> header file. */
+#undef HAVE_SYS_TYPES_H
+
+/* Define to 1 if you have <sys/wait.h> that is POSIX.1 compatible. */
+#undef HAVE_SYS_WAIT_H
+
+/* Define to 1 if you have the <tgmath.h> header file. */
+#undef HAVE_TGMATH_H
+
+/* Define to 1 if you have the <tiffio.h> header file. */
+#undef HAVE_TIFFIO_H
+
+/* Define to 1 if you have the <unicall.h> header file. */
+#undef HAVE_UNICALL_H
+
+/* Define to 1 if you have the <unistd.h> header file. */
+#undef HAVE_UNISTD_H
+
+/* Define to 1 if you have the `vprintf' function. */
+#undef HAVE_VPRINTF
+
+/* Define to 1 if you have the <X11/X.h> header file. */
+#undef HAVE_X11_X_H
+
+/* Name of package */
+#undef PACKAGE
+
+/* Define to the address where bug reports for this package should be sent. */
+#undef PACKAGE_BUGREPORT
+
+/* Define to the full name of this package. */
+#undef PACKAGE_NAME
+
+/* Define to the full name and version of this package. */
+#undef PACKAGE_STRING
+
+/* Define to the one symbol short name of this package. */
+#undef PACKAGE_TARNAME
+
+/* Define to the version of this package. */
+#undef PACKAGE_VERSION
+
+/* Define as the return type of signal handlers (`int' or `void'). */
+#undef RETSIGTYPE
+
+/* Define to the type of arg 1 for `select'. */
+#undef SELECT_TYPE_ARG1
+
+/* Define to the type of args 2, 3 and 4 for `select'. */
+#undef SELECT_TYPE_ARG234
+
+/* Define to the type of arg 5 for `select'. */
+#undef SELECT_TYPE_ARG5
+
+/* Define to 1 if you have the ANSI C header files. */
+#undef STDC_HEADERS
+
+/* Define to 1 if you can safely include both <sys/time.h> and <time.h>. */
+#undef TIME_WITH_SYS_TIME
+
+/* Version number of package */
+#undef VERSION
+
+/* Enable fixed point processing, where possible, instead of floating point */
+#undef VOIPCODECS_USE_FIXED_POINT
+
+/* Define to empty if `const' does not conform to ANSI C. */
+#undef const
+
+/* Define to `__inline__' or `__inline' if that's what the C compiler
+   calls it, or to nothing if 'inline' is not supported under any name.  */
+#ifndef __cplusplus
+#undef inline
+#endif
+
+/* Define to rpl_malloc if the replacement function should be used. */
+#undef malloc
+
+/* Define to rpl_realloc if the replacement function should be used. */
+#undef realloc
+
+/* Define to empty if the keyword `volatile' does not work. Warning: valid
+   code using `volatile' can become incorrect without. Disable with care. */
+#undef volatile

Added: freeswitch/trunk/libs/voipcodecs/configure.ac
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/configure.ac	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,340 @@
+#
+# VoIPcodecs - a series of DSP components for telephony
+#
+# configure.ac -- Process this file with autoconf to produce configure
+#
+# This program is free software; you can redistribute it and/or modify
+# it under the terms of the GNU General Public License version 2, as
+# published by the Free Software Foundation.
+#
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+# GNU General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, write to the Free Software
+# Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+#
+# $Id: configure.ac,v 1.38 2007/11/10 11:14:56 steveu Exp $
+
+# @start 1
+
+AC_INIT
+
+AC_DEFUN([AX_COMPILER_VENDOR],
+[
+AC_CACHE_CHECK([for _AC_LANG compiler vendor], ax_cv_[]_AC_LANG_ABBREV[]_compiler_vendor,
+ [ax_cv_[]_AC_LANG_ABBREV[]_compiler_vendor=unknown
+  # note: don't check for GCC first, since some other compilers define __GNUC__
+  for ventest in intel:__ICC,__ECC,__INTEL_COMPILER ibm:__xlc__,__xlC__,__IBMC__,__IBMCPP__ gnu:__GNUC__ sun:__SUNPRO_C,__SUNPRO_CC hp:__HP_cc,__HP_aCC dec:__DECC,__DECCXX,__DECC_VER,__DECCXX_VER borland:__BORLANDC__,__TURBOC__ comeau:__COMO__ cray:_CRAYC kai:__KCC lcc:__LCC__ metrowerks:__MWERKS__ sgi:__sgi,sgi microsoft:_MSC_VER watcom:__WATCOMC__ portland:__PGI; do
+    vencpp="defined("`echo $ventest | cut -d: -f2 | sed 's/,/) || defined(/g'`")"
+    AC_COMPILE_IFELSE([AC_LANG_PROGRAM(,[
+#if !($vencpp)
+      thisisanerror;
+#endif
+])], [ax_cv_]_AC_LANG_ABBREV[_compiler_vendor=`echo $ventest | cut -d: -f1`; break])
+  done
+ ])
+])
+
+VOIPCODECS_MAJOR_VERSION=0
+VOIPCODECS_MINOR_VERSION=0
+VOIPCODECS_MICRO_VERSION=1
+
+VOIPCODECS_LT_CURRENT=0
+VOIPCODECS_LT_REVISION=2
+VOIPCODECS_LT_AGE=0
+
+VERSION=$VOIPCODECS_MAJOR_VERSION.$VOIPCODECS_MINOR_VERSION.$VOIPCODECS_MICRO_VERSION
+PACKAGE=voipcodecs
+
+AC_SUBST(VOIPCODECS_LT_CURRENT)
+AC_SUBST(VOIPCODECS_LT_REVISION)
+AC_SUBST(VOIPCODECS_LT_AGE)
+
+AC_CONFIG_SRCDIR([src/g711.c])
+AC_CONFIG_AUX_DIR(config)
+AC_CONFIG_HEADERS([src/config.h:config-h.in])
+AM_INIT_AUTOMAKE($PACKAGE, $VERSION)
+
+AC_CANONICAL_HOST
+AC_CANONICAL_BUILD
+AC_PROG_CC
+AC_PROG_CXX
+AC_PROG_GCC_TRADITIONAL
+AC_PROG_LIBTOOL
+AC_LANG([C])
+
+AX_COMPILER_VENDOR
+
+if test "${build}" != "${host}"
+then
+  # If we are doing a Canadian Cross, in which the host and build systems
+  # are not the same, we set reasonable default values for the tools.
+
+  CC=${CC-${host_alias}-gcc}
+  CFLAGS=${CFLAGS-"-g -O2"}
+  CXX=${CXX-${host_alias}-c++}
+  CXXFLAGS=${CXXFLAGS-"-g -O2"}
+  CC_FOR_BUILD=${CC_FOR_BUILD-gcc}
+else
+  # Set reasonable default values for some tools even if not Canadian.
+  # Of course, these are different reasonable default values, originally
+  # specified directly in the Makefile.
+  # We don't export, so that autoconf can do its job.
+  # Note that all these settings are above the fragment inclusion point
+  # in Makefile.in, so can still be overridden by fragments.
+  # This is all going to change when we autoconfiscate...
+  CC_FOR_BUILD="\$(CC)"
+  AC_PROG_CC
+
+  # We must set the default linker to the linker used by gcc for the correct
+  # operation of libtool.  If LD is not defined and we are using gcc, try to
+  # set the LD default to the ld used by gcc.
+  if test -z "$LD"
+  then
+    if test "$GCC" = yes
+    then
+      case $build in
+      *-*-mingw*)
+        gcc_prog_ld=`$CC -print-prog-name=ld 2>&1 | tr -d '\015'` ;;
+      *)
+        gcc_prog_ld=`$CC -print-prog-name=ld 2>&1` ;;
+      esac
+      case $gcc_prog_ld in
+      # Accept absolute paths.
+      [[\\/]* | [A-Za-z]:[\\/]*)]
+        LD="$gcc_prog_ld" ;;
+      esac
+    fi
+  fi
+
+  CXX=${CXX-"c++"}
+  CFLAGS=${CFLAGS-"-g -O2"}
+  CXXFLAGS=${CXXFLAGS-"-g -O2"}
+fi
+
+AC_C_CONST
+AC_C_INLINE
+AC_C_VOLATILE
+
+AC_CHECK_TYPES(long long)
+AC_CHECK_TYPES(long double)
+
+AC_TYPE_SIGNAL
+
+AC_ARG_ENABLE(doc,          [  --enable-doc         Build the documentation])
+AC_ARG_ENABLE(tests,        [  --enable-tests       Build the test programs])
+AC_ARG_ENABLE(itutests,     [  --enable-itutests    Build TIFF test files for some ITU test images])
+AC_ARG_ENABLE(mmx,          [  --enable-mmx         Enable MMX support])
+AC_ARG_ENABLE(sse,          [  --enable-sse         Enable SSE support])
+AC_ARG_ENABLE(fixed_point,  [  --enable-fixed-point Enable fixed point support])
+
+AC_FUNC_ERROR_AT_LINE
+AC_FUNC_VPRINTF
+AC_FUNC_MALLOC
+AC_FUNC_MEMCMP
+AC_FUNC_REALLOC
+AC_FUNC_SELECT_ARGTYPES
+
+AC_CHECK_FUNCS([memmove])
+AC_CHECK_FUNCS([memset])
+AC_CHECK_FUNCS([select])
+AC_CHECK_FUNCS([strcasecmp])
+AC_CHECK_FUNCS([strchr])
+AC_CHECK_FUNCS([strdup])
+AC_CHECK_FUNCS([strerror])
+AC_CHECK_FUNCS([strstr])
+AC_CHECK_FUNCS([strtol])
+AC_CHECK_FUNCS([gettimeofday])
+
+AC_HEADER_STDC
+AC_HEADER_SYS_WAIT
+AC_HEADER_TIME
+
+# Check for header files.
+AC_CHECK_HEADERS([socket.h])
+AC_CHECK_HEADERS([inttypes.h], [INSERT_INTTYPES_HEADER="#include <inttypes.h>"])
+AC_CHECK_HEADERS([stdint.h], [INSERT_STDINT_HEADER="#include <stdint.h>"])
+AC_CHECK_HEADERS([unistd.h])
+AC_CHECK_HEADERS([stdlib.h])
+AC_CHECK_HEADERS([string.h])
+AC_CHECK_HEADERS([strings.h])
+AC_CHECK_HEADERS([malloc.h])
+AC_CHECK_HEADERS([tgmath.h], [INSERT_TGMATH_HEADER="#include <tgmath.h>"])
+AC_CHECK_HEADERS([math.h], [INSERT_MATH_HEADER="#include <math.h>"])
+AC_CHECK_HEADERS([float.h])
+AC_CHECK_HEADERS([fcntl.h])
+AC_CHECK_HEADERS([sys/time.h])
+AC_CHECK_HEADERS([sys/select.h])
+AC_CHECK_HEADERS([sys/ioctl.h])
+AC_CHECK_HEADERS([sys/fcntl.h])
+AC_CHECK_HEADERS([audiofile.h])
+AC_CHECK_HEADERS([fftw.h])
+AC_CHECK_HEADERS([fftw3.h])
+AC_CHECK_HEADERS([tiffio.h])
+AC_CHECK_HEADERS([pthread.h])
+AC_CHECK_HEADERS([unicall.h])
+if test "${build}" == "${host}"
+then
+  AC_CHECK_HEADERS([X11/X.h])
+fi
+
+# Determine XML2 include path
+AC_MSG_CHECKING(for libxml/xmlmemory.h)
+
+# Can we include headers using system include dirs?
+AC_TRY_COMPILE([#include <libxml/xmlmemory.h>], [int a = 1;],
+    XML2_INCLUDE=" ",
+    XML2_INCLUDE=
+)
+
+# Hunt through several possible directories to find the includes for libxml2
+if test "x$XML2_INCLUDE" = "x"; then
+    old_CPPFLAGS="$CPPFLAGS"
+    for i in $xml2_include_dir /usr/include /usr/local/include /usr/include/libxml2 /usr/local/include/libxml2 ; do
+        CPPFLAGS="$old_CPPFLAGS -I$i"
+        AC_TRY_COMPILE([#include <libxml/xmlmemory.h>], [int a = 1;],
+            XML2_INCLUDE="-I$i",
+            XML2_INCLUDE=
+        )
+        if test "x$XML2_INCLUDE" != "x"; then
+            break;
+        fi
+    done
+    CPPFLAGS="$old_CPPFLAGS $XML2_INCLUDE"
+fi
+
+AC_CHECK_HEADERS([libxml/xmlmemory.h])
+AC_CHECK_HEADERS([libxml/parser.h])
+AC_CHECK_HEADERS([libxml/xinclude.h])
+
+AC_LANG([C++])
+AC_CHECK_HEADERS([FL/Fl.H])
+AC_CHECK_HEADERS([FL/Fl_Overlay_Window.H])
+AC_CHECK_HEADERS([FL/Fl_Light_Button.H])
+AC_CHECK_HEADERS([FL/fl_draw.H])
+AC_CHECK_HEADERS([FL/Fl_Cartesian.H], [], [], [],[[#include <FL/Fl.H>
+]])
+AC_CHECK_HEADERS([FL/Fl_Audio_Meter.H], [], [], [],[[#include <FL/Fl.H>
+]])
+
+if test "${build}" == "${host}"
+then
+    case "${host}" in
+    x86_64-*)
+        AC_CHECK_FILE([${prefix}/lib64], libdir='$(exec_prefix)/lib64')
+        ;;
+    esac
+fi
+
+AC_LANG([C])
+
+if test "${build}" == "${host}"
+then
+    case "${host}" in
+    x86_64-*)
+        # X86_64 Linux machines may have both 64 bit and 32 bit libraries. We need to choose the right set
+        AC_CHECK_FILE([/usr/X11R6/lib64], [TESTLIBS="$TESTLIBS -L/usr/X11R6/lib64"], AC_CHECK_FILE([/usr/X11R6/lib], [TESTLIBS="$TESTLIBS -L/usr/X11R6/lib"]))
+        ;;
+    esac
+fi
+
+# Checks for libraries.
+AC_CHECK_LIB([Xft], [XftFontOpen], TESTLIBS="$TESTLIBS -lXft",, $TESTLIBS)
+AC_CHECK_LIB([Xext], [XextCreateExtension], TESTLIBS="$TESTLIBS -lXext",, $TESTLIBS)
+AC_CHECK_LIB([X11], [XOpenDisplay], TESTLIBS="$TESTLIBS -lX11",, $TESTLIBS)
+AC_CHECK_LIB([tiff], [TIFFOpen], , AC_MSG_ERROR("Can't build without libtiff (does your system require a libtiff-devel package?)"), -lm)
+AC_CHECK_LIB([m], [cos])
+AC_CHECK_LIB([m], [pow])
+AC_CHECK_LIB([m], [sqrt])
+AC_CHECK_LIB([xml2], [xmlParseFile], [AC_DEFINE([HAVE_LIBXML2], [1], [Define to 1 if you have the 'libxml2' library (-lxml2).]) TESTLIBS="$TESTLIBS -lxml2"])
+if test -n "$enable_tests" ; then
+    AC_LANG([C++])
+    AC_CHECK_LIB([fltk], [main], TESTLIBS="$TESTLIBS -lfltk -lsupc++")
+    AC_CHECK_LIB([fltk_cartesian], [main], TESTLIBS="-lfltk_cartesian $TESTLIBS")
+    AC_CHECK_LIB([fltk_audio_meter], [main], TESTLIBS="-lfltk_audio_meter $TESTLIBS")
+    AC_LANG([C])
+    AC_CHECK_LIB([audiofile], [afOpenFile], TESTLIBS="$TESTLIBS -laudiofile", AC_MSG_ERROR("Can't make tests without libaudiofile (does your system require a libaudiofile-devel package?)"))
+    AC_CHECK_LIB([fftw3], [fftw_plan_dft_1d], TESTLIBS="$TESTLIBS -lfftw3")
+    AC_CHECK_LIB([fftw], [fftw_create_plan], TESTLIBS="$TESTLIBS -lfftw")
+    AC_CHECK_LIB([pthread], [pthread_attr_init], TESTLIBS="$TESTLIBS -lpthread")
+    AC_CHECK_LIB([dl], [dlopen], TESTLIBS="$TESTLIBS -ldl")
+    AC_CHECK_LIB([unicall], [uc_start], TESTLIBS="$TESTLIBS -lunicall",, -ltiff -ldl)
+fi
+
+case "${ax_cv_c_compiler_vendor}" in
+gnu)
+    COMP_VENDOR_CFLAGS="-std=gnu99 -ffast-math -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
+    if test "$enable_sse" = "yes" ; then
+        COMP_VENDOR_CFLAGS="-msse $COMP_VENDOR_CFLAGS"
+    fi
+    if test "$enable_mmx" = "yes" ; then
+        COMP_VENDOR_CFLAGS="-mmmx $COMP_VENDOR_CFLAGS"
+    fi
+    ;;
+*)
+    COMP_VENDOR_CFLAGS="-std=c99 -Wall -Wunused-variable -Wwrite-strings -Wstrict-prototypes -Wmissing-prototypes"
+    ;;
+esac
+
+COMP_VENDOR_CFLAGS="-DNDEBUG $COMP_VENDOR_CFLAGS"
+
+AM_CONDITIONAL([COND_DOC], [test "$enable_doc" = yes])
+AM_CONDITIONAL([COND_TESTS], [test "$enable_tests" = yes])
+AM_CONDITIONAL([COND_ITUTESTS], [test "$enable_itutests" = yes])
+AM_CONDITIONAL([COND_MMX], [test "$enable_mmx" = yes])
+AM_CONDITIONAL([COND_SSE], [test "$enable_sse" = yes])
+if test "$enable_fixed_point" = "yes" ; then
+    AC_DEFINE([VOIPCODECS_USE_FIXED_POINT], [1], [Enable fixed point processing, where possible, instead of floating point])
+    VOIPCODECS_USE_FIXED_POINT="#define VOIPCODECS_USE_FIXED_POINT 1"
+else
+#
+# So far we deal with the embedded ARM, Blackfin, MIPS, TI DSP and XScale processors as
+# things which lack fast hardware floating point.
+#
+# Other candidates would be the small embedded Power PCs.
+#
+    case $basic_machine in
+	      arc | arm | arm[bl]e | arme[lb] | armv[2345] | armv[345][lb] \
+	    | bfin \
+	    | mips | mipsbe | mipseb | mipsel | mipsle \
+	    | tic54x | c54x* | tic55x | c55x* | tic6x | c6x* \
+	    | xscale | xscalee[bl] \
+	    | arm-*  | armbe-* | armle-* | armeb-* | armv*-* \
+	    | bfin-* \
+	    | mips-* | mipsbe-* | mipseb-* | mipsel-* | mipsle-* \
+	    | tic30-* | tic4x-* | tic54x-* | tic55x-* | tic6x-* | tic80-* \
+	    | xscale-* | xscalee[bl]-* )
+        AC_DEFINE([VOIPCODECS_USE_FIXED_POINT], [1], [Enable fixed point processing, where possible, instead of floating point])
+        VOIPCODECS_USE_FIXED_POINT="#define VOIPCODECS_USE_FIXED_POINT 1"
+        ;;
+    *)
+        VOIPCODECS_USE_FIXED_POINT="#undef VOIPCODECS_USE_FIXED_POINT"
+        ;;
+    esac
+fi
+
+AC_SUBST(CC_FOR_BUILD)
+AC_SUBST(COMP_VENDOR_CFLAGS)
+AC_SUBST(TESTLIBS)
+AC_SUBST(VOIPCODECS_USE_FIXED_POINT)
+AC_SUBST(INSERT_INTTYPES_HEADER)
+AC_SUBST(INSERT_STDINT_HEADER)
+AC_SUBST(INSERT_TGMATH_HEADER)
+AC_SUBST(INSERT_MATH_HEADER)
+
+AC_CONFIG_FILES([Makefile
+                 doc/Makefile
+                 etsitests/Makefile
+                 itutests/Makefile
+                 localtests/Makefile
+                 src/Makefile
+                 src/voipcodecs.h
+                 tests/Makefile
+                 voipcodecs.spec])
+
+AC_OUTPUT
+
+# @end 1

Added: freeswitch/trunk/libs/voipcodecs/debian/changelog
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/debian/changelog	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,6 @@
+voipcodecs (0.0.1) unstable; urgency=low
+
+  [ Steve Underwood ]
+  * Begun
+
+ -- Steve Underwood <steveu@coppice.org>  Thu,  7 Feb 2008 09:53:06 +0300

Added: freeswitch/trunk/libs/voipcodecs/debian/compat
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/debian/compat	Thu Feb  7 17:00:10 2008
@@ -0,0 +1 @@
+4

Added: freeswitch/trunk/libs/voipcodecs/debian/control
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/debian/control	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,42 @@
+Source: voipcodecs
+Section: libs
+Priority: optional
+Maintainer: Debian VoIP Team <pkg-voip-maintainers@lists.alioth.debian.org>
+Uploaders: Jose Carlos Garcia Sogo <jsogo@debian.org>, Kilian Krause <kilian@debian.org>, Santiago Garcia Mantinan <manty@debian.org>, Mark Purcell <msp@debian.org>, Tzafrir Cohen <tzafrir.cohen@xorcom.com>, Santiago Ruano Rincón <santiago@debian.org>
+Build-Depends: debhelper (>= 4.0.0), libtiff4-dev, libjpeg62-dev, dpatch, doxygen, autotools-dev
+Standards-Version: 3.7.2
+XS-Vcs-Svn: svn://svn.debian.org/pkg-voip/
+XS-Vcs-Browser: http://svn.debian.org/wsvn/pkg-voip/
+
+Package: libvoipcodecs3
+Architecture: any
+Depends: ${shlibs:Depends}
+Conflicts: libvoipcodecs0, libvoipcodecs1, libvoipcodecs2
+Description: Telephony signal processing library
+ This is a low-level signal processing library that modulate and demodulate
+ signals commonly used in telephony, such as the "noise" generated by a
+ fax modem or DTMF touchpad.
+ .
+ This package contains the shared library.
+
+Package: libvoipcodecs-dev
+Section: libdevel
+Architecture: any
+Depends: libvoipcodecs3 (= ${Source-Version}), libtiff4-dev, libjpeg62-dev
+Description: Telephony signal processing library
+ This is a low-level signal processing library that modulate and demodulate
+ signals commonly used in telephony, such as the "noise" generated by a
+ fax modem or DTMF touchpad.
+ .
+ This package contains the static library and development headers.
+ .
+ Homepage: http://www.soft-switch.org/
+
+Package: libvoipcodecs-doc
+Section: doc
+Architecture: all
+Description: Documentation for the voipcodecs signal processing library
+ This package contains the online API in HTML for the libvoipcodecs, a low
+ level signal processing library that modulate and demodulate siignals
+ commonly used in telephony, such as the "noise" generated by a fax
+ modem or DTMF touchpad.

Added: freeswitch/trunk/libs/voipcodecs/debian/copyright
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/debian/copyright	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,25 @@
+This package was debianized by Steve Underwood <steveu@coppice.org> on
+Thu,  7 Feb 2008 15:22:58 +0100.
+
+It was downloaded from http://soft-switch.org/downloads/voipcodecs/
+
+Copyright: Steve Underwood <steveu@coppice.org>
+
+License:
+
+   This package is free software; you can redistribute it and/or modify
+   it under the terms of the GNU General Public License as published by
+   the Free Software Foundation; version 2 dated June, 1991.
+
+   This package is distributed in the hope that it will be useful,
+   but WITHOUT ANY WARRANTY; without even the implied warranty of
+   MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+   GNU General Public License for more details.
+
+   You should have received a copy of the GNU General Public License along with
+   this package; if not, write to the Free Software Foundation, Inc., 
+   51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA.
+
+
+On Debian systems, the complete text of the GNU General
+Public License can be found in `/usr/share/common-licenses/GPL'.

Added: freeswitch/trunk/libs/voipcodecs/debian/libvoipcodecs-dev.install
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/debian/libvoipcodecs-dev.install	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,4 @@
+debian/tmp/usr/include
+debian/tmp/usr/lib/libvoipcodecs.so
+debian/tmp/usr/lib/libvoipcodecs.la
+debian/tmp/usr/lib/libvoipcodecs.a

Added: freeswitch/trunk/libs/voipcodecs/debian/libvoipcodecs-doc.install
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/debian/libvoipcodecs-doc.install	Thu Feb  7 17:00:10 2008
@@ -0,0 +1 @@
+doc/api/html usr/share/doc/libvoipcodecs-doc/api/

Added: freeswitch/trunk/libs/voipcodecs/debian/libvoipcodecs.install
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/debian/libvoipcodecs.install	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,2 @@
+debian/tmp/usr/lib/libvoipcodecs.so.0.*
+debian/tmp/usr/lib/libvoipcodecs.so.0

Added: freeswitch/trunk/libs/voipcodecs/debian/rules
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/debian/rules	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,106 @@
+#!/usr/bin/make -f
+
+# Uncomment this to turn on verbose mode.
+#export DH_VERBOSE=1
+
+DEB_HOST_GNU_TYPE   ?= $(shell dpkg-architecture -qDEB_HOST_GNU_TYPE)
+DEB_BUILD_GNU_TYPE  ?= $(shell dpkg-architecture -qDEB_BUILD_GNU_TYPE)
+
+DEBVERSION:=$(shell head -n 1 debian/changelog \
+		    | sed -e 's/^[^(]*(\([^)]*\)).*/\1/')
+ORIGTARVER:=$(shell echo $(DEBVERSION) | sed -e 's/^.*://' -e 's/-[0-9.]*$$//')# -e 's/.dfsg$$//' -e 's/~//')
+
+UPVERSION:=$(shell echo $(ORIGTARVER) | tr -d '~')
+
+FILENAME := voipcodecs_$(ORIGTARVER).orig.tar.gz
+FULLNAME := voipcodecs-$(UPVERSION)
+URL := http://soft-switch.org/downloads/voipcodecs/voipcodecs-$(UPVERSION).tgz
+
+CFLAGS = -Wall -g
+
+ifneq (,$(findstring noopt,$(DEB_BUILD_OPTIONS)))
+	CFLAGS += -O0
+else
+	CFLAGS += -O2
+endif
+
+include /usr/share/dpatch/dpatch.make
+
+
+autotools: patch-stamp
+	ln -s /usr/share/misc/config.sub config.sub
+	ln -s /usr/share/misc/config.guess config.guess
+	touch autotools
+
+config.status: autotools configure
+	dh_testdir
+	CFLAGS="$(CFLAGS)" ./configure \
+	  --host=$(DEB_HOST_GNU_TYPE) \
+	  --build=$(DEB_BUILD_GNU_TYPE) \
+	  --prefix=/usr \
+	  --mandir=\$${prefix}/share/man \
+	  --infodir=\$${prefix}/share/info \
+	  --enable-doc
+
+build: build-stamp
+
+build-stamp:  config.status
+	dh_testdir
+	$(MAKE)
+	touch build-stamp
+
+clean: clean-patched unpatch
+clean-patched:
+	dh_testdir
+	dh_testroot
+	rm -f build-stamp autotools
+	-$(MAKE) distclean
+
+	-$(RM) -f config.sub
+	-$(RM) -f config.guess
+
+	dh_clean 
+
+install: build-stamp
+	dh_testdir
+	dh_testroot
+	dh_clean -k 
+	$(MAKE) install DESTDIR=$(CURDIR)/debian/tmp
+
+binary-indep: build-stamp install
+	dh_testdir -i
+	dh_testroot -i
+	dh_installchangelogs -i ChangeLog
+	dh_installdocs -i DueDiligence
+	dh_install -i
+	dh_compress -i
+	dh_fixperms -i
+	dh_installdeb -i
+	dh_gencontrol -i
+	dh_md5sums -i
+	dh_builddeb -i
+
+binary-arch: build-stamp install
+	dh_testdir -a
+	dh_testroot -a
+	dh_installchangelogs -a ChangeLog
+	dh_installdocs -a DueDiligence
+	dh_install -a
+	dh_strip -a
+	dh_compress -a
+	dh_fixperms -a
+	dh_makeshlibs -a
+	dh_installdeb -a
+	dh_shlibdeps -a
+	dh_gencontrol -a
+	dh_md5sums -a
+	dh_builddeb -a
+
+get-orig-source:
+	-@@dh_testdir
+	@@[ -d ../tarballs/. ]||mkdir -p ../tarballs
+	@@echo Downloading $(FILENAME) from $(URL) ...
+	@@wget -N -nv -T10 -t3 -O ../tarballs/$(FILENAME) $(URL)
+
+binary: binary-indep binary-arch
+.PHONY: build clean binary-indep binary-arch binary install patch unpatch

Added: freeswitch/trunk/libs/voipcodecs/debian/watch
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/debian/watch	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,7 @@
+# See uscan(1) for format
+
+# Compulsory line, this is a version 3 file
+version=3
+
+# <Webpage URL> <string match>
+http://soft-switch.org/downloads/voipcodecs/ voipcodecs-(.*)\.tgz debian svn-upgrade

Added: freeswitch/trunk/libs/voipcodecs/doc/Makefile.am
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/doc/Makefile.am	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,29 @@
+##
+## VoIPcodecs - a series of DSP components for telephony
+##
+## Makefile.am -- Process this file with automake to produce Makefile.in
+##
+## This program is free software; you can redistribute it and/or modify
+## it under the terms of the GNU General Public License version 2, as
+## published by the Free Software Foundation.
+##
+## This program is distributed in the hope that it will be useful,
+## but WITHOUT ANY WARRANTY; without even the implied warranty of
+## MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+## GNU General Public License for more details.
+##
+## You should have received a copy of the GNU General Public License
+## along with this program; if not, write to the Free Software
+## Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+##
+## $Id: Makefile.am,v 1.7 2007/12/20 15:06:22 steveu Exp $
+
+MAINTAINERCLEANFILES = Makefile.in
+
+EXTRA_DIST = css.css \
+             wrapper.xsl
+
+all: doxydocs
+
+doxydocs:
+	doxygen libvoipcodecs-doxygen

Added: freeswitch/trunk/libs/voipcodecs/doc/css.css
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/doc/css.css	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,564 @@
+body {
+	background-image: url("../images/weave.jpg");
+	font-family: Verdana, Arial, Helvetica, Sans-serif;
+	color: black;
+	margin-right: 20px;
+	margin-left: 20px;
+}
+
+h1 {
+	text-align: center;
+}
+
+h2 {
+	font-family: Verdana, Arial, Helvetica, Sans-serif;
+	border-color: #c00000;
+	color : black;
+	margin-top: 0.8em;
+	border-style: solid;
+	border-width: 0px 0px 3px 0.5em;
+	line-height : 130%;
+}
+
+h3 {
+	font-family: Verdana, Arial, Helvetica, Sans-serif;
+	border-color: #f02020;
+	color : black;
+	border-width: 0px 0px 2px 0.5em;
+	border-style: solid;
+	margin-right: 20%;
+	line-height : 130%;
+}
+caption {
+	font-weight: bold
+}
+a.qindex {}
+a.qindexRef {}
+a.el {
+	text-decoration: none;
+	font-weight: bold
+}
+a.elRef {
+	font-weight: bold
+}
+a.code {
+	text-decoration: none;
+	font-weight: normal;
+	color: #4444ee
+}
+a.codeRef {
+	font-weight: normal;
+	color: #4444ee
+}
+a:hover {
+	text-decoration: none;
+	background-color: #f2f2ff
+}
+dl.el {
+	margin-left: -1cm
+}
+div.fragment {
+	width: 100%;
+	border: none;
+	background-color: #eeeeee
+}
+div.ah {
+	background-color: black;
+	font-weight: bold;
+	color: #ffffff;
+	margin-bottom: 3px;
+	margin-top: 3px
+}
+
+td {
+	font-family: Verdana, Arial, Helvetica, Sans-serif;
+	font-weight: bold;
+}
+
+.navheader {
+	font-family: Verdana, Arial, Helvetica, Sans-serif;
+	background-color: #B2B2ff;
+	font-weight: bold;
+}
+
+.navfooter {
+	font-family: Verdana, Arial, Helvetica, Sans-serif;
+	background-color: #B2B2ff;
+	font-weight: bold;
+}
+
+table.menu {
+	background-color: #000066;
+	font-weight: bold;
+	text-align: center;
+	width: 100%; 
+}
+
+tr.menu {
+	background-color: #ccffff;
+	font-weight: bold;
+	text-align: center;
+}
+td.menu {
+	background-color: #f2e0d0;
+	font-weight: bold;
+	text-align: center;
+}
+
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Added: freeswitch/trunk/libs/voipcodecs/doc/libvoipcodecs-doxygen
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/doc/libvoipcodecs-doxygen	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,1229 @@
+# Doxyfile 1.4.4
+
+# This file describes the settings to be used by the documentation system
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+#---------------------------------------------------------------------------
+# Project related configuration options
+#---------------------------------------------------------------------------
+
+# The PROJECT_NAME tag is a single word (or a sequence of words surrounded 
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+
+PROJECT_NAME           = libvoipcodecs
+
+# The PROJECT_NUMBER tag can be used to enter a project or revision number. 
+# This could be handy for archiving the generated documentation or 
+# if some version control system is used.
+
+PROJECT_NUMBER         = 0.0.1
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+
+CREATE_SUBDIRS         = NO
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+
+OUTPUT_LANGUAGE        = English
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+# documentation blocks found inside the body of a function. 
+# If set to NO (the default) these blocks will be appended to the 
+# function's detailed documentation block.
+
+HIDE_IN_BODY_DOCS      = NO
+
+# The INTERNAL_DOCS tag determines if documentation 
+# that is typed after a \internal command is included. If the tag is set 
+# to NO (the default) then the documentation will be excluded. 
+# Set it to YES to include the internal documentation.
+
+INTERNAL_DOCS          = NO
+
+# If the CASE_SENSE_NAMES tag is set to NO then Doxygen will only generate 
+# file names in lower-case letters. If set to YES upper-case letters are also 
+# allowed. This is useful if you have classes or files whose names only differ 
+# in case and if your file system supports case sensitive file names. Windows 
+# and Mac users are advised to set this option to NO.
+
+CASE_SENSE_NAMES       = YES
+
+# If the HIDE_SCOPE_NAMES tag is set to NO (the default) then Doxygen 
+# will show members with their full class and namespace scopes in the 
+# documentation. If set to YES the scope will be hidden.
+
+HIDE_SCOPE_NAMES       = NO
+
+# If the SHOW_INCLUDE_FILES tag is set to YES (the default) then Doxygen 
+# will put a list of the files that are included by a file in the documentation 
+# of that file.
+
+SHOW_INCLUDE_FILES     = YES
+
+# If the INLINE_INFO tag is set to YES (the default) then a tag [inline] 
+# is inserted in the documentation for inline members.
+
+INLINE_INFO            = YES
+
+# If the SORT_MEMBER_DOCS tag is set to YES (the default) then doxygen 
+# will sort the (detailed) documentation of file and class members 
+# alphabetically by member name. If set to NO the members will appear in 
+# declaration order.
+
+SORT_MEMBER_DOCS       = YES
+
+# If the SORT_BRIEF_DOCS tag is set to YES then doxygen will sort the 
+# brief documentation of file, namespace and class members alphabetically 
+# by member name. If set to NO (the default) the members will appear in 
+# declaration order.
+
+SORT_BRIEF_DOCS        = NO
+
+# If the SORT_BY_SCOPE_NAME tag is set to YES, the class list will be 
+# sorted by fully-qualified names, including namespaces. If set to 
+# NO (the default), the class list will be sorted only by class name, 
+# not including the namespace part. 
+# Note: This option is not very useful if HIDE_SCOPE_NAMES is set to YES.
+# Note: This option applies only to the class list, not to the 
+# alphabetical list.
+
+SORT_BY_SCOPE_NAME     = NO
+
+# The GENERATE_TODOLIST tag can be used to enable (YES) or 
+# disable (NO) the todo list. This list is created by putting \todo 
+# commands in the documentation.
+
+GENERATE_TODOLIST      = YES
+
+# The GENERATE_TESTLIST tag can be used to enable (YES) or 
+# disable (NO) the test list. This list is created by putting \test 
+# commands in the documentation.
+
+GENERATE_TESTLIST      = YES
+
+# The GENERATE_BUGLIST tag can be used to enable (YES) or 
+# disable (NO) the bug list. This list is created by putting \bug 
+# commands in the documentation.
+
+GENERATE_BUGLIST       = YES
+
+# The GENERATE_DEPRECATEDLIST tag can be used to enable (YES) or 
+# disable (NO) the deprecated list. This list is created by putting 
+# \deprecated commands in the documentation.
+
+GENERATE_DEPRECATEDLIST= YES
+
+# The ENABLED_SECTIONS tag can be used to enable conditional 
+# documentation sections, marked by \if sectionname ... \endif.
+
+ENABLED_SECTIONS       = 
+
+# The MAX_INITIALIZER_LINES tag determines the maximum number of lines 
+# the initial value of a variable or define consists of for it to appear in 
+# the documentation. If the initializer consists of more lines than specified 
+# here it will be hidden. Use a value of 0 to hide initializers completely. 
+# The appearance of the initializer of individual variables and defines in the 
+# documentation can be controlled using \showinitializer or \hideinitializer 
+# command in the documentation regardless of this setting.
+
+MAX_INITIALIZER_LINES  = 30
+
+# Set the SHOW_USED_FILES tag to NO to disable the list of files generated 
+# at the bottom of the documentation of classes and structs. If set to YES the 
+# list will mention the files that were used to generate the documentation.
+
+SHOW_USED_FILES        = YES
+
+# If the sources in your project are distributed over multiple directories 
+# then setting the SHOW_DIRECTORIES tag to YES will show the directory hierarchy 
+# in the documentation. The default is YES.
+
+SHOW_DIRECTORIES       = YES
+
+# The FILE_VERSION_FILTER tag can be used to specify a program or script that 
+# doxygen should invoke to get the current version for each file (typically from the 
+# version control system). Doxygen will invoke the program by executing (via 
+# popen()) the command <command> <input-file>, where <command> is the value of 
+# the FILE_VERSION_FILTER tag, and <input-file> is the name of an input file 
+# provided by doxygen. Whatever the progam writes to standard output 
+# is used as the file version. See the manual for examples.
+
+FILE_VERSION_FILTER    = 
+
+#---------------------------------------------------------------------------
+# configuration options related to warning and progress messages
+#---------------------------------------------------------------------------
+
+# The QUIET tag can be used to turn on/off the messages that are generated 
+# by doxygen. Possible values are YES and NO. If left blank NO is used.
+
+QUIET                  = NO
+
+# The WARNINGS tag can be used to turn on/off the warning messages that are 
+# generated by doxygen. Possible values are YES and NO. If left blank 
+# NO is used.
+
+WARNINGS               = YES
+
+# If WARN_IF_UNDOCUMENTED is set to YES, then doxygen will generate warnings 
+# for undocumented members. If EXTRACT_ALL is set to YES then this flag will 
+# automatically be disabled.
+
+WARN_IF_UNDOCUMENTED   = YES
+
+# If WARN_IF_DOC_ERROR is set to YES, doxygen will generate warnings for 
+# potential errors in the documentation, such as not documenting some 
+# parameters in a documented function, or documenting parameters that 
+# don't exist or using markup commands wrongly.
+
+WARN_IF_DOC_ERROR      = YES
+
+# This WARN_NO_PARAMDOC option can be abled to get warnings for 
+# functions that are documented, but have no documentation for their parameters 
+# or return value. If set to NO (the default) doxygen will only warn about 
+# wrong or incomplete parameter documentation, but not about the absence of 
+# documentation.
+
+WARN_NO_PARAMDOC       = NO
+
+# The WARN_FORMAT tag determines the format of the warning messages that 
+# doxygen can produce. The string should contain the $file, $line, and $text 
+# tags, which will be replaced by the file and line number from which the 
+# warning originated and the warning text. Optionally the format may contain 
+# $version, which will be replaced by the version of the file (if it could 
+# be obtained via FILE_VERSION_FILTER)
+
+WARN_FORMAT            = 
+
+# The WARN_LOGFILE tag can be used to specify a file to which warning 
+# and error messages should be written. If left blank the output is written 
+# to stderr.
+
+WARN_LOGFILE           = 
+
+#---------------------------------------------------------------------------
+# configuration options related to the input files
+#---------------------------------------------------------------------------
+
+# The INPUT tag can be used to specify the files and/or directories that contain 
+# documented source files. You may enter file names like "myfile.cpp" or 
+# directories like "/usr/src/myproject". Separate the files or directories 
+# with spaces.
+
+INPUT                  = ..
+
+# If the value of the INPUT tag contains directories, you can use the 
+# FILE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp 
+# and *.h) to filter out the source-files in the directories. If left 
+# blank the following patterns are tested: 
+# *.c *.cc *.cxx *.cpp *.c++ *.java *.ii *.ixx *.ipp *.i++ *.inl *.h *.hh *.hxx 
+# *.hpp *.h++ *.idl *.odl *.cs *.php *.php3 *.inc *.m *.mm
+
+FILE_PATTERNS          = *.c \
+                         *.h
+
+# The RECURSIVE tag can be used to turn specify whether or not subdirectories 
+# should be searched for input files as well. Possible values are YES and NO. 
+# If left blank NO is used.
+
+RECURSIVE              = YES
+
+# The EXCLUDE tag can be used to specify files and/or directories that should 
+# excluded from the INPUT source files. This way you can easily exclude a 
+# subdirectory from a directory tree whose root is specified with the INPUT tag.
+
+EXCLUDE                = 
+
+# The EXCLUDE_SYMLINKS tag can be used select whether or not files or 
+# directories that are symbolic links (a Unix filesystem feature) are excluded 
+# from the input.
+
+EXCLUDE_SYMLINKS       = NO
+
+# If the value of the INPUT tag contains directories, you can use the 
+# EXCLUDE_PATTERNS tag to specify one or more wildcard patterns to exclude 
+# certain files from those directories. Note that the wildcards are matched 
+# against the file with absolute path, so to exclude all test directories 
+# for example use the pattern */test/*
+
+EXCLUDE_PATTERNS       = 
+
+# The EXAMPLE_PATH tag can be used to specify one or more files or 
+# directories that contain example code fragments that are included (see 
+# the \include command).
+
+EXAMPLE_PATH           = 
+
+# If the value of the EXAMPLE_PATH tag contains directories, you can use the 
+# EXAMPLE_PATTERNS tag to specify one or more wildcard pattern (like *.cpp 
+# and *.h) to filter out the source-files in the directories. If left 
+# blank all files are included.
+
+EXAMPLE_PATTERNS       = 
+
+# If the EXAMPLE_RECURSIVE tag is set to YES then subdirectories will be 
+# searched for input files to be used with the \include or \dontinclude 
+# commands irrespective of the value of the RECURSIVE tag. 
+# Possible values are YES and NO. If left blank NO is used.
+
+EXAMPLE_RECURSIVE      = NO
+
+# The IMAGE_PATH tag can be used to specify one or more files or 
+# directories that contain image that are included in the documentation (see 
+# the \image command).
+
+IMAGE_PATH             = 
+
+# The INPUT_FILTER tag can be used to specify a program that doxygen should 
+# invoke to filter for each input file. Doxygen will invoke the filter program 
+# by executing (via popen()) the command <filter> <input-file>, where <filter> 
+# is the value of the INPUT_FILTER tag, and <input-file> is the name of an 
+# input file. Doxygen will then use the output that the filter program writes 
+# to standard output.  If FILTER_PATTERNS is specified, this tag will be 
+# ignored.
+
+INPUT_FILTER           = 
+
+# The FILTER_PATTERNS tag can be used to specify filters on a per file pattern 
+# basis.  Doxygen will compare the file name with each pattern and apply the 
+# filter if there is a match.  The filters are a list of the form: 
+# pattern=filter (like *.cpp=my_cpp_filter). See INPUT_FILTER for further 
+# info on how filters are used. If FILTER_PATTERNS is empty, INPUT_FILTER 
+# is applied to all files.
+
+FILTER_PATTERNS        = 
+
+# If the FILTER_SOURCE_FILES tag is set to YES, the input filter (if set using 
+# INPUT_FILTER) will be used to filter the input files when producing source 
+# files to browse (i.e. when SOURCE_BROWSER is set to YES).
+
+FILTER_SOURCE_FILES    = NO
+
+#---------------------------------------------------------------------------
+# configuration options related to source browsing
+#---------------------------------------------------------------------------
+
+# If the SOURCE_BROWSER tag is set to YES then a list of source files will 
+# be generated. Documented entities will be cross-referenced with these sources. 
+# Note: To get rid of all source code in the generated output, make sure also 
+# VERBATIM_HEADERS is set to NO.
+
+SOURCE_BROWSER         = NO
+
+# Setting the INLINE_SOURCES tag to YES will include the body 
+# of functions and classes directly in the documentation.
+
+INLINE_SOURCES         = NO
+
+# Setting the STRIP_CODE_COMMENTS tag to YES (the default) will instruct 
+# doxygen to hide any special comment blocks from generated source code 
+# fragments. Normal C and C++ comments will always remain visible.
+
+STRIP_CODE_COMMENTS    = NO
+
+# If the REFERENCED_BY_RELATION tag is set to YES (the default) 
+# then for each documented function all documented 
+# functions referencing it will be listed.
+
+REFERENCED_BY_RELATION = YES
+
+# If the REFERENCES_RELATION tag is set to YES (the default) 
+# then for each documented function all documented entities 
+# called/used by that function will be listed.
+
+REFERENCES_RELATION    = YES
+
+# If the USE_HTAGS tag is set to YES then the references to source code 
+# will point to the HTML generated by the htags(1) tool instead of doxygen 
+# built-in source browser. The htags tool is part of GNU's global source 
+# tagging system (see http://www.gnu.org/software/global/global.html). You 
+# will need version 4.8.6 or higher.
+
+USE_HTAGS              = NO
+
+# If the VERBATIM_HEADERS tag is set to YES (the default) then Doxygen 
+# will generate a verbatim copy of the header file for each class for 
+# which an include is specified. Set to NO to disable this.
+
+VERBATIM_HEADERS       = YES
+
+#---------------------------------------------------------------------------
+# configuration options related to the alphabetical class index
+#---------------------------------------------------------------------------
+
+# If the ALPHABETICAL_INDEX tag is set to YES, an alphabetical index 
+# of all compounds will be generated. Enable this if the project 
+# contains a lot of classes, structs, unions or interfaces.
+
+ALPHABETICAL_INDEX     = YES
+
+# If the alphabetical index is enabled (see ALPHABETICAL_INDEX) then 
+# the COLS_IN_ALPHA_INDEX tag can be used to specify the number of columns 
+# in which this list will be split (can be a number in the range [1..20])
+
+COLS_IN_ALPHA_INDEX    = 2
+
+# In case all classes in a project start with a common prefix, all 
+# classes will be put under the same header in the alphabetical index. 
+# The IGNORE_PREFIX tag can be used to specify one or more prefixes that 
+# should be ignored while generating the index headers.
+
+IGNORE_PREFIX          = 
+
+#---------------------------------------------------------------------------
+# configuration options related to the HTML output
+#---------------------------------------------------------------------------
+
+# If the GENERATE_HTML tag is set to YES (the default) Doxygen will 
+# generate HTML output.
+
+GENERATE_HTML          = YES
+
+# The HTML_OUTPUT tag is used to specify where the HTML docs will be put. 
+# If a relative path is entered the value of OUTPUT_DIRECTORY will be 
+# put in front of it. If left blank `html' will be used as the default path.
+
+HTML_OUTPUT            = 
+
+# The HTML_FILE_EXTENSION tag can be used to specify the file extension for 
+# each generated HTML page (for example: .htm,.php,.asp). If it is left blank 
+# doxygen will generate files with .html extension.
+
+HTML_FILE_EXTENSION    = .html
+
+# The HTML_HEADER tag can be used to specify a personal HTML header for 
+# each generated HTML page. If it is left blank doxygen will generate a 
+# standard header.
+
+HTML_HEADER            = 
+
+# The HTML_FOOTER tag can be used to specify a personal HTML footer for 
+# each generated HTML page. If it is left blank doxygen will generate a 
+# standard footer.
+
+HTML_FOOTER            = 
+
+# The HTML_STYLESHEET tag can be used to specify a user-defined cascading 
+# style sheet that is used by each HTML page. It can be used to 
+# fine-tune the look of the HTML output. If the tag is left blank doxygen 
+# will generate a default style sheet. Note that doxygen will try to copy 
+# the style sheet file to the HTML output directory, so don't put your own 
+# stylesheet in the HTML output directory as well, or it will be erased!
+
+HTML_STYLESHEET        = css.css
+
+# If the HTML_ALIGN_MEMBERS tag is set to YES, the members of classes, 
+# files or namespaces will be aligned in HTML using tables. If set to 
+# NO a bullet list will be used.
+
+HTML_ALIGN_MEMBERS     = YES
+
+# If the GENERATE_HTMLHELP tag is set to YES, additional index files 
+# will be generated that can be used as input for tools like the 
+# Microsoft HTML help workshop to generate a compressed HTML help file (.chm) 
+# of the generated HTML documentation.
+
+GENERATE_HTMLHELP      = NO
+
+# If the GENERATE_HTMLHELP tag is set to YES, the CHM_FILE tag can 
+# be used to specify the file name of the resulting .chm file. You 
+# can add a path in front of the file if the result should not be 
+# written to the html output directory.
+
+CHM_FILE               = 
+
+# If the GENERATE_HTMLHELP tag is set to YES, the HHC_LOCATION tag can 
+# be used to specify the location (absolute path including file name) of 
+# the HTML help compiler (hhc.exe). If non-empty doxygen will try to run 
+# the HTML help compiler on the generated index.hhp.
+
+HHC_LOCATION           = 
+
+# If the GENERATE_HTMLHELP tag is set to YES, the GENERATE_CHI flag 
+# controls if a separate .chi index file is generated (YES) or that 
+# it should be included in the master .chm file (NO).
+
+GENERATE_CHI           = NO
+
+# If the GENERATE_HTMLHELP tag is set to YES, the BINARY_TOC flag 
+# controls whether a binary table of contents is generated (YES) or a 
+# normal table of contents (NO) in the .chm file.
+
+BINARY_TOC             = NO
+
+# The TOC_EXPAND flag can be set to YES to add extra items for group members 
+# to the contents of the HTML help documentation and to the tree view.
+
+TOC_EXPAND             = NO
+
+# The DISABLE_INDEX tag can be used to turn on/off the condensed index at 
+# top of each HTML page. The value NO (the default) enables the index and 
+# the value YES disables it.
+
+DISABLE_INDEX          = NO
+
+# This tag can be used to set the number of enum values (range [1..20]) 
+# that doxygen will group on one line in the generated HTML documentation.
+
+ENUM_VALUES_PER_LINE   = 4
+
+# If the GENERATE_TREEVIEW tag is set to YES, a side panel will be
+# generated containing a tree-like index structure (just like the one that 
+# is generated for HTML Help). For this to work a browser that supports 
+# JavaScript, DHTML, CSS and frames is required (for instance Mozilla 1.0+, 
+# Netscape 6.0+, Internet explorer 5.0+, or Konqueror). Windows users are 
+# probably better off using the HTML help feature.
+
+GENERATE_TREEVIEW      = YES
+
+# If the treeview is enabled (see GENERATE_TREEVIEW) then this tag can be 
+# used to set the initial width (in pixels) of the frame in which the tree 
+# is shown.
+
+TREEVIEW_WIDTH         = 250
+
+#---------------------------------------------------------------------------
+# configuration options related to the LaTeX output
+#---------------------------------------------------------------------------
+
+# If the GENERATE_LATEX tag is set to YES (the default) Doxygen will 
+# generate Latex output.
+
+GENERATE_LATEX         = NO
+
+# The LATEX_OUTPUT tag is used to specify where the LaTeX docs will be put. 
+# If a relative path is entered the value of OUTPUT_DIRECTORY will be 
+# put in front of it. If left blank `latex' will be used as the default path.
+
+LATEX_OUTPUT           = 
+
+# The LATEX_CMD_NAME tag can be used to specify the LaTeX command name to be 
+# invoked. If left blank `latex' will be used as the default command name.
+
+LATEX_CMD_NAME         = latex
+
+# The MAKEINDEX_CMD_NAME tag can be used to specify the command name to 
+# generate index for LaTeX. If left blank `makeindex' will be used as the 
+# default command name.
+
+MAKEINDEX_CMD_NAME     = makeindex
+
+# If the COMPACT_LATEX tag is set to YES Doxygen generates more compact 
+# LaTeX documents. This may be useful for small projects and may help to 
+# save some trees in general.
+
+COMPACT_LATEX          = NO
+
+# The PAPER_TYPE tag can be used to set the paper type that is used 
+# by the printer. Possible values are: a4, a4wide, letter, legal and 
+# executive. If left blank a4wide will be used.
+
+PAPER_TYPE             = a4wide
+
+# The EXTRA_PACKAGES tag can be to specify one or more names of LaTeX 
+# packages that should be included in the LaTeX output.
+
+EXTRA_PACKAGES         = 
+
+# The LATEX_HEADER tag can be used to specify a personal LaTeX header for 
+# the generated latex document. The header should contain everything until 
+# the first chapter. If it is left blank doxygen will generate a 
+# standard header. Notice: only use this tag if you know what you are doing!
+
+LATEX_HEADER           = 
+
+# If the PDF_HYPERLINKS tag is set to YES, the LaTeX that is generated 
+# is prepared for conversion to pdf (using ps2pdf). The pdf file will 
+# contain links (just like the HTML output) instead of page references 
+# This makes the output suitable for online browsing using a pdf viewer.
+
+PDF_HYPERLINKS         = NO
+
+# If the USE_PDFLATEX tag is set to YES, pdflatex will be used instead of 
+# plain latex in the generated Makefile. Set this option to YES to get a 
+# higher quality PDF documentation.
+
+USE_PDFLATEX           = NO
+
+# If the LATEX_BATCHMODE tag is set to YES, doxygen will add the \\batchmode. 
+# command to the generated LaTeX files. This will instruct LaTeX to keep 
+# running if errors occur, instead of asking the user for help. 
+# This option is also used when generating formulas in HTML.
+
+LATEX_BATCHMODE        = NO
+
+# If LATEX_HIDE_INDICES is set to YES then doxygen will not 
+# include the index chapters (such as File Index, Compound Index, etc.) 
+# in the output.
+
+LATEX_HIDE_INDICES     = NO
+
+#---------------------------------------------------------------------------
+# configuration options related to the RTF output
+#---------------------------------------------------------------------------
+
+# If the GENERATE_RTF tag is set to YES Doxygen will generate RTF output 
+# The RTF output is optimized for Word 97 and may not look very pretty with 
+# other RTF readers or editors.
+
+GENERATE_RTF           = NO
+
+# The RTF_OUTPUT tag is used to specify where the RTF docs will be put. 
+# If a relative path is entered the value of OUTPUT_DIRECTORY will be 
+# put in front of it. If left blank `rtf' will be used as the default path.
+
+RTF_OUTPUT             = 
+
+# If the COMPACT_RTF tag is set to YES Doxygen generates more compact 
+# RTF documents. This may be useful for small projects and may help to 
+# save some trees in general.
+
+COMPACT_RTF            = NO
+
+# If the RTF_HYPERLINKS tag is set to YES, the RTF that is generated 
+# will contain hyperlink fields. The RTF file will 
+# contain links (just like the HTML output) instead of page references. 
+# This makes the output suitable for online browsing using WORD or other 
+# programs which support those fields. 
+# Note: wordpad (write) and others do not support links.
+
+RTF_HYPERLINKS         = NO
+
+# Load stylesheet definitions from file. Syntax is similar to doxygen's 
+# config file, i.e. a series of assignments. You only have to provide 
+# replacements, missing definitions are set to their default value.
+
+RTF_STYLESHEET_FILE    = 
+
+# Set optional variables used in the generation of an rtf document. 
+# Syntax is similar to doxygen's config file.
+
+RTF_EXTENSIONS_FILE    = 
+
+#---------------------------------------------------------------------------
+# configuration options related to the man page output
+#---------------------------------------------------------------------------
+
+# If the GENERATE_MAN tag is set to YES (the default) Doxygen will 
+# generate man pages
+
+GENERATE_MAN           = NO
+
+# The MAN_OUTPUT tag is used to specify where the man pages will be put. 
+# If a relative path is entered the value of OUTPUT_DIRECTORY will be 
+# put in front of it. If left blank `man' will be used as the default path.
+
+MAN_OUTPUT             = 
+
+# The MAN_EXTENSION tag determines the extension that is added to 
+# the generated man pages (default is the subroutine's section .3)
+
+MAN_EXTENSION          = 
+
+# If the MAN_LINKS tag is set to YES and Doxygen generates man output, 
+# then it will generate one additional man file for each entity 
+# documented in the real man page(s). These additional files 
+# only source the real man page, but without them the man command 
+# would be unable to find the correct page. The default is NO.
+
+MAN_LINKS              = NO
+
+#---------------------------------------------------------------------------
+# configuration options related to the XML output
+#---------------------------------------------------------------------------
+
+# If the GENERATE_XML tag is set to YES Doxygen will 
+# generate an XML file that captures the structure of 
+# the code including all documentation.
+
+GENERATE_XML           = NO
+
+# The XML_OUTPUT tag is used to specify where the XML pages will be put. 
+# If a relative path is entered the value of OUTPUT_DIRECTORY will be 
+# put in front of it. If left blank `xml' will be used as the default path.
+
+XML_OUTPUT             = xml
+
+# The XML_SCHEMA tag can be used to specify an XML schema, 
+# which can be used by a validating XML parser to check the 
+# syntax of the XML files.
+
+XML_SCHEMA             = 
+
+# The XML_DTD tag can be used to specify an XML DTD, 
+# which can be used by a validating XML parser to check the 
+# syntax of the XML files.
+
+XML_DTD                = 
+
+# If the XML_PROGRAMLISTING tag is set to YES Doxygen will 
+# dump the program listings (including syntax highlighting 
+# and cross-referencing information) to the XML output. Note that 
+# enabling this will significantly increase the size of the XML output.
+
+XML_PROGRAMLISTING     = YES
+
+#---------------------------------------------------------------------------
+# configuration options for the AutoGen Definitions output
+#---------------------------------------------------------------------------
+
+# If the GENERATE_AUTOGEN_DEF tag is set to YES Doxygen will 
+# generate an AutoGen Definitions (see autogen.sf.net) file 
+# that captures the structure of the code including all 
+# documentation. Note that this feature is still experimental 
+# and incomplete at the moment.
+
+GENERATE_AUTOGEN_DEF   = NO
+
+#---------------------------------------------------------------------------
+# configuration options related to the Perl module output
+#---------------------------------------------------------------------------
+
+# If the GENERATE_PERLMOD tag is set to YES Doxygen will 
+# generate a Perl module file that captures the structure of 
+# the code including all documentation. Note that this 
+# feature is still experimental and incomplete at the 
+# moment.
+
+GENERATE_PERLMOD       = NO
+
+# If the PERLMOD_LATEX tag is set to YES Doxygen will generate 
+# the necessary Makefile rules, Perl scripts and LaTeX code to be able 
+# to generate PDF and DVI output from the Perl module output.
+
+PERLMOD_LATEX          = NO
+
+# If the PERLMOD_PRETTY tag is set to YES the Perl module output will be 
+# nicely formatted so it can be parsed by a human reader.  This is useful 
+# if you want to understand what is going on.  On the other hand, if this 
+# tag is set to NO the size of the Perl module output will be much smaller 
+# and Perl will parse it just the same.
+
+PERLMOD_PRETTY         = YES
+
+# The names of the make variables in the generated doxyrules.make file 
+# are prefixed with the string contained in PERLMOD_MAKEVAR_PREFIX. 
+# This is useful so different doxyrules.make files included by the same 
+# Makefile don't overwrite each other's variables.
+
+PERLMOD_MAKEVAR_PREFIX = 
+
+#---------------------------------------------------------------------------
+# Configuration options related to the preprocessor   
+#---------------------------------------------------------------------------
+
+# If the ENABLE_PREPROCESSING tag is set to YES (the default) Doxygen will 
+# evaluate all C-preprocessor directives found in the sources and include 
+# files.
+
+ENABLE_PREPROCESSING   = YES
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+# If the MACRO_EXPANSION tag is set to YES Doxygen will expand all macro 
+# names in the source code. If set to NO (the default) only conditional 
+# compilation will be performed. Macro expansion can be done in a controlled 
+# way by setting EXPAND_ONLY_PREDEF to YES.
+
+MACRO_EXPANSION        = NO
+
+# If the EXPAND_ONLY_PREDEF and MACRO_EXPANSION tags are both set to YES 
+# then the macro expansion is limited to the macros specified with the 
+# PREDEFINED and EXPAND_AS_PREDEFINED tags.
+
+EXPAND_ONLY_PREDEF     = NO
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+# If the SEARCH_INCLUDES tag is set to YES (the default) the includes files 
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+SEARCH_INCLUDES        = YES
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+# The INCLUDE_PATH tag can be used to specify one or more directories that 
+# contain include files that are not input files but should be processed by 
+# the preprocessor.
+
+INCLUDE_PATH           = 
+
+# You can use the INCLUDE_FILE_PATTERNS tag to specify one or more wildcard 
+# patterns (like *.h and *.hpp) to filter out the header-files in the 
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+INCLUDE_FILE_PATTERNS  = 
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+# The PREDEFINED tag can be used to specify one or more macro names that 
+# are defined before the preprocessor is started (similar to the -D option of 
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+# omitted =1 is assumed. To prevent a macro definition from being 
+# undefined via #undef or recursively expanded use the := operator 
+# instead of the = operator.
+
+PREDEFINED             = 
+
+# If the MACRO_EXPANSION and EXPAND_ONLY_PREDEF tags are set to YES then 
+# this tag can be used to specify a list of macro names that should be expanded. 
+# The macro definition that is found in the sources will be used. 
+# Use the PREDEFINED tag if you want to use a different macro definition.
+
+EXPAND_AS_DEFINED      = 
+
+# If the SKIP_FUNCTION_MACROS tag is set to YES (the default) then 
+# doxygen's preprocessor will remove all function-like macros that are alone 
+# on a line, have an all uppercase name, and do not end with a semicolon. Such 
+# function macros are typically used for boiler-plate code, and will confuse 
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+
+SKIP_FUNCTION_MACROS   = YES
+
+#---------------------------------------------------------------------------
+# Configuration::additions related to external references   
+#---------------------------------------------------------------------------
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+# The TAGFILES option can be used to specify one or more tagfiles. 
+# Optionally an initial location of the external documentation 
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+#   TAGFILES = file1 file2 ... 
+# Adding location for the tag files is done as follows: 
+#   TAGFILES = file1=loc1 "file2 = loc2" ... 
+# where "loc1" and "loc2" can be relative or absolute paths or 
+# URLs. If a location is present for each tag, the installdox tool 
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+# Note that each tag file must have a unique name
+# (where the name does NOT include the path)
+# If a tag file is not located in the directory in which doxygen 
+# is run, you must also specify the path to the tagfile here.
+
+TAGFILES               = 
+
+# When a file name is specified after GENERATE_TAGFILE, doxygen will create 
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+
+GENERATE_TAGFILE       = 
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+# If the ALLEXTERNALS tag is set to YES all external classes will be listed 
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+# will be listed.
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+EXTERNAL_GROUPS        = YES
+
+# The PERL_PATH should be the absolute path and name of the perl script 
+# interpreter (i.e. the result of `which perl').
+
+PERL_PATH              = /usr/bin/perl
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+#---------------------------------------------------------------------------
+# Configuration options related to the dot tool   
+#---------------------------------------------------------------------------
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+# If the CLASS_DIAGRAMS tag is set to YES (the default) Doxygen will 
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+# fallback. It is recommended to install and use dot, since it yields more 
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+
+CLASS_DIAGRAMS         = YES
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+# If set to YES, the inheritance and collaboration graphs will hide 
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+# If you set the HAVE_DOT tag to YES then doxygen will assume the dot tool is 
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+HAVE_DOT               = NO
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+# If the CLASS_GRAPH and HAVE_DOT tags are set to YES then doxygen 
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+
+CLASS_GRAPH            = YES
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+COLLABORATION_GRAPH    = YES
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+# If the GROUP_GRAPHS and HAVE_DOT tags are set to YES then doxygen 
+# will generate a graph for groups, showing the direct groups dependencies
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+# If the ENABLE_PREPROCESSING, SEARCH_INCLUDES, INCLUDE_GRAPH, and HAVE_DOT 
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+INCLUDED_BY_GRAPH      = YES
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+# If the CALL_GRAPH and HAVE_DOT tags are set to YES then doxygen will 
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+
+CALL_GRAPH             = NO
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+# If the GRAPHICAL_HIERARCHY and HAVE_DOT tags are set to YES then doxygen 
+# will graphical hierarchy of all classes instead of a textual one.
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+GRAPHICAL_HIERARCHY    = YES
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+DIRECTORY_GRAPH        = YES
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+DOT_IMAGE_FORMAT       = png
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+# found. If left blank, it is assumed the dot tool can be found in the path.
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+DOT_PATH               = 
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+# contain dot files that are included in the documentation (see the 
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+DOTFILE_DIRS           = 
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+# (in pixels) of the graphs generated by dot. If a graph becomes larger than 
+# this value, doxygen will try to truncate the graph, so that it fits within 
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+MAX_DOT_GRAPH_WIDTH    = 1024
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+# this value, doxygen will try to truncate the graph, so that it fits within 
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+MAX_DOT_GRAPH_HEIGHT   = 1024
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+# graphs generated by dot. A depth value of 3 means that only nodes reachable 
+# from the root by following a path via at most 3 edges will be shown. Nodes 
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+# image dimensions are not sufficient to fit the graph (see MAX_DOT_GRAPH_WIDTH 
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+
+MAX_DOT_GRAPH_DEPTH    = 0
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+# Set the DOT_TRANSPARENT tag to YES to generate images with a transparent 
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+# Warning: Depending on the platform used, enabling this option may lead to 
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+# read).
+
+DOT_TRANSPARENT        = NO
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+# Set the DOT_MULTI_TARGETS tag to YES allow dot to generate multiple output 
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+# support this, this feature is disabled by default.
+
+DOT_MULTI_TARGETS      = NO
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+# If the GENERATE_LEGEND tag is set to YES (the default) Doxygen will 
+# generate a legend page explaining the meaning of the various boxes and 
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+
+GENERATE_LEGEND        = YES
+
+# If the DOT_CLEANUP tag is set to YES (the default) Doxygen will 
+# remove the intermediate dot files that are used to generate 
+# the various graphs.
+
+DOT_CLEANUP            = YES
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+# The SEARCHENGINE tag specifies whether or not a search engine should be 
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+SEARCHENGINE           = NO

Added: freeswitch/trunk/libs/voipcodecs/doc/wrapper.xsl
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/doc/wrapper.xsl	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,5 @@
+<xsl:stylesheet xmlns:xsl="http://www.w3.org/1999/XSL/Transform"
+                version='1.0'>
+  <xsl:import href="http://docbook.sourceforge.net/release/xsl/current/xhtml/chunk.xsl"/>
+  <xsl:param name="html.stylesheet">css.css</xsl:param>
+</xsl:stylesheet>
\ No newline at end of file

Added: freeswitch/trunk/libs/voipcodecs/etsitests/Makefile.am
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/etsitests/Makefile.am	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,27 @@
+##
+## VoIPcodecs - a series of DSP components for telephony
+##
+## Makefile.am -- Process this file with automake to produce Makefile.in
+##
+## This program is free software; you can redistribute it and/or modify
+## it under the terms of the GNU General Public License version 2, as
+## published by the Free Software Foundation.
+##
+## This program is distributed in the hope that it will be useful,
+## but WITHOUT ANY WARRANTY; without even the implied warranty of
+## MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+## GNU General Public License for more details.
+##
+## You should have received a copy of the GNU General Public License
+## along with this program; if not, write to the Free Software
+## Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+##
+## $Id: Makefile.am,v 1.1 2008/02/07 10:49:02 steveu Exp $
+
+SUBDIRS =
+
+DIST_SUBDIRS =
+
+all: 
+
+clean:

Added: freeswitch/trunk/libs/voipcodecs/itutests/Makefile.am
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/itutests/Makefile.am	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,28 @@
+##
+## SpanDSP - a series of DSP components for telephony
+##
+## Makefile.am -- Process this file with automake to produce Makefile.in
+##
+## This program is free software; you can redistribute it and/or modify
+## it under the terms of the GNU General Public License version 2, as
+## published by the Free Software Foundation.
+##
+## This program is distributed in the hope that it will be useful,
+## but WITHOUT ANY WARRANTY; without even the implied warranty of
+## MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+## GNU General Public License for more details.
+##
+## You should have received a copy of the GNU General Public License
+## along with this program; if not, write to the Free Software
+## Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+##
+## $Id: Makefile.am,v 1.5 2007/02/10 11:54:57 steveu Exp $
+
+SUBDIRS =
+
+DIST_SUBDIRS =
+
+all: 
+
+clean:
+

Added: freeswitch/trunk/libs/voipcodecs/libvoipcodecs.vcproj
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/libvoipcodecs.vcproj	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,657 @@
+<?xml version="1.0" encoding="Windows-1252"?>
+<VisualStudioProject
+	ProjectType="Visual C++"
+	Version="8.00"
+	Name="libvoipcodecs"
+	ProjectGUID="{CF70F278-3364-4395-A2E1-23501C9B8AD2}"
+	RootNamespace="libvoipcodecs"
+	Keyword="Win32Proj"
+	>
+	<Platforms>
+		<Platform
+			Name="Win32"
+		/>
+	</Platforms>
+	<ToolFiles>
+	</ToolFiles>
+	<Configurations>
+		<Configuration
+			Name="Debug|Win32"
+			OutputDirectory="$(SolutionDir)$(ConfigurationName)"
+			IntermediateDirectory="$(ConfigurationName)"
+			ConfigurationType="2"
+			CharacterSet="1"
+			>
+			<Tool
+				Name="VCPreBuildEventTool"
+			/>
+			<Tool
+				Name="VCCustomBuildTool"
+			/>
+			<Tool
+				Name="VCXMLDataGeneratorTool"
+			/>
+			<Tool
+				Name="VCWebServiceProxyGeneratorTool"
+			/>
+			<Tool
+				Name="VCMIDLTool"
+			/>
+			<Tool
+				Name="VCCLCompilerTool"
+				Optimization="0"
+				AdditionalIncludeDirectories="src; src\voipcodecs; src\msvc"
+				PreprocessorDefinitions="WIN32;_DEBUG;_WINDOWS;_USRDLL;LIBvoipcodecs_EXPORTS;_CRT_SECURE_NO_DEPRECATE;_CRT_NONSTDC_NO_DEPRECATE"
+				MinimalRebuild="true"
+				BasicRuntimeChecks="3"
+				RuntimeLibrary="3"
+				UsePrecompiledHeader="0"
+				WarningLevel="4"
+				Detect64BitPortabilityProblems="true"
+				DebugInformationFormat="4"
+			/>
+			<Tool
+				Name="VCManagedResourceCompilerTool"
+			/>
+			<Tool
+				Name="VCResourceCompilerTool"
+			/>
+			<Tool
+				Name="VCPreLinkEventTool"
+			/>
+			<Tool
+				Name="VCLinkerTool"
+				LinkIncremental="2"
+				ModuleDefinitionFile="src/msvc/voipcodecs.def"
+				GenerateDebugInformation="true"
+				SubSystem="2"
+				TargetMachine="1"
+			/>
+			<Tool
+				Name="VCALinkTool"
+			/>
+			<Tool
+				Name="VCManifestTool"
+			/>
+			<Tool
+				Name="VCXDCMakeTool"
+			/>
+			<Tool
+				Name="VCBscMakeTool"
+			/>
+			<Tool
+				Name="VCFxCopTool"
+			/>
+			<Tool
+				Name="VCAppVerifierTool"
+			/>
+			<Tool
+				Name="VCWebDeploymentTool"
+			/>
+			<Tool
+				Name="VCPostBuildEventTool"
+			/>
+		</Configuration>
+		<Configuration
+			Name="Release|Win32"
+			OutputDirectory="$(SolutionDir)$(ConfigurationName)"
+			IntermediateDirectory="$(ConfigurationName)"
+			ConfigurationType="2"
+			CharacterSet="1"
+			WholeProgramOptimization="1"
+			>
+			<Tool
+				Name="VCPreBuildEventTool"
+			/>
+			<Tool
+				Name="VCCustomBuildTool"
+			/>
+			<Tool
+				Name="VCXMLDataGeneratorTool"
+			/>
+			<Tool
+				Name="VCWebServiceProxyGeneratorTool"
+			/>
+			<Tool
+				Name="VCMIDLTool"
+			/>
+			<Tool
+				Name="VCCLCompilerTool"
+				AdditionalIncludeDirectories="src; src\voipcodecs; src\msvc"
+				PreprocessorDefinitions="WIN32;NDEBUG;_WINDOWS;_USRDLL;LIBvoipcodecs_EXPORTS;_CRT_SECURE_NO_DEPRECATE;_CRT_NONSTDC_NO_DEPRECATE"
+				RuntimeLibrary="2"
+				UsePrecompiledHeader="0"
+				WarningLevel="4"
+				Detect64BitPortabilityProblems="true"
+				DebugInformationFormat="3"
+			/>
+			<Tool
+				Name="VCManagedResourceCompilerTool"
+			/>
+			<Tool
+				Name="VCResourceCompilerTool"
+			/>
+			<Tool
+				Name="VCPreLinkEventTool"
+			/>
+			<Tool
+				Name="VCLinkerTool"
+				LinkIncremental="1"
+				ModuleDefinitionFile="src/msvc/voipcodecs.def"
+				GenerateDebugInformation="true"
+				SubSystem="2"
+				OptimizeReferences="2"
+				EnableCOMDATFolding="2"
+				TargetMachine="1"
+			/>
+			<Tool
+				Name="VCALinkTool"
+			/>
+			<Tool
+				Name="VCManifestTool"
+			/>
+			<Tool
+				Name="VCXDCMakeTool"
+			/>
+			<Tool
+				Name="VCBscMakeTool"
+			/>
+			<Tool
+				Name="VCFxCopTool"
+			/>
+			<Tool
+				Name="VCAppVerifierTool"
+			/>
+			<Tool
+				Name="VCWebDeploymentTool"
+			/>
+			<Tool
+				Name="VCPostBuildEventTool"
+			/>
+		</Configuration>
+	</Configurations>
+	<References>
+	</References>
+	<Files>
+		<Filter
+			Name="Source Files"
+			Filter="cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx"
+			UniqueIdentifier="{4FC737F1-C7A5-4376-A066-2A32D752A2FF}"
+			>
+			<File
+				RelativePath=".\src\adsi.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\async.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\awgn.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\bert.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\complex_dds.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\complex_filters.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\dds.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\ec_disable_tone.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\echo.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\fax.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\fsk.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\g722_decode.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\g722_encode.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\g726.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\msvc\gettimeofday.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\hdlc.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\ima_adpcm.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\logging.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\modem_echo.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\noise.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\oki_adpcm.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\playout.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\plc.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\power_meter.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\queue.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\schedule.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\sig_tone.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\super_tone_rx.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\super_tone_tx.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\t30.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\t31.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\t35.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\t38_gateway.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\t38_ifppacket.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\t38_terminal.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\t4.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\testcpuid.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\time_scale.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\tone_detect.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\tone_generate.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\v17rx.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\v17tx.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\v22bis_rx.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\v22bis_tx.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\v27ter_rx.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\v27ter_tx.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\v29rx.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\v29tx.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\v42.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\v42bis.c"
+				>
+			</File>
+			<File
+				RelativePath=".\src\v8.c"
+				>
+			</File>
+		</Filter>
+		<Filter
+			Name="Header Files"
+			Filter="h;hpp;hxx;hm;inl;inc;xsd"
+			UniqueIdentifier="{93995380-89BD-4b04-88EB-625FBE52EBFB}"
+			>
+			<File
+				RelativePath=".\src\voipcodecs\adsi.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\alaw_ulaw.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\arctan2.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\async.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\awgn.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\bert.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\biquad.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\complex.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\complex_filters.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\dc_restore.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\dds.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\ec_disable_detector.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\ec_disable_tone.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\echo.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\fax.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\faxfont.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\fir.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\fsk.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\g168models.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\g722.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\g726.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\hdlc.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\ima_adpcm.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\msvc\inttypes.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\logging.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\modem_echo.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\noise.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\oki_adpcm.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\oss.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\playout.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\plc.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\power_meter.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\queue.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\schedule.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\sig_tone.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\super_tone_rx.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\super_tone_tx.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\t30.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\t30_fcf.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\t31.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\t35.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\t38.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\t4.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\t4states.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\telephony.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\msvc\tgmath.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\msvc\sys\time.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\time_scale.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\timing.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\tone_detect.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\tone_generate.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\msvc\unistd.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\v17rx.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\v17tx.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\v22bis.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\v27ter_rx.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\v27ter_tx.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\v29rx.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\v29tx.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\v42.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\v42bis.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\v8.h"
+				>
+			</File>
+			<File
+				RelativePath=".\src\voipcodecs\vector.h"
+				>
+			</File>
+		</Filter>
+		<Filter
+			Name="Resource Files"
+			Filter="rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx;tiff;tif;png;wav"
+			UniqueIdentifier="{67DA6AB6-F800-4c08-8B7A-83BB121AAD01}"
+			>
+		</Filter>
+	</Files>
+	<Globals>
+	</Globals>
+</VisualStudioProject>

Added: freeswitch/trunk/libs/voipcodecs/localtests/Makefile.am
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/localtests/Makefile.am	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,32 @@
+##
+## VoIPcodecs - a series of DSP components for telephony
+##
+## Makefile.am -- Process this file with automake to produce Makefile.in
+##
+## This program is free software; you can redistribute it and/or modify
+## it under the terms of the GNU General Public License version 2, as
+## published by the Free Software Foundation.
+##
+## This program is distributed in the hope that it will be useful,
+## but WITHOUT ANY WARRANTY; without even the implied warranty of
+## MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+## GNU General Public License for more details.
+##
+## You should have received a copy of the GNU General Public License
+## along with this program; if not, write to the Free Software
+## Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+##
+## $Id: Makefile.am,v 1.1 2008/02/07 10:49:02 steveu Exp $
+
+SUBDIRS =
+
+DIST_SUBDIRS =
+
+EXTRA_DIST = dam9_lpc55.wav \
+             dam9.wav \
+             short_nb_voice.wav \
+             short_wb_voice.wav
+
+all: 
+
+clean:

Added: freeswitch/trunk/libs/voipcodecs/localtests/dam9.wav
==============================================================================
Binary file. No diff available.

Added: freeswitch/trunk/libs/voipcodecs/localtests/dam9_lpc55.wav
==============================================================================
Binary file. No diff available.

Added: freeswitch/trunk/libs/voipcodecs/localtests/short_nb_voice.wav
==============================================================================
Binary file. No diff available.

Added: freeswitch/trunk/libs/voipcodecs/localtests/short_wb_voice.wav
==============================================================================
Binary file. No diff available.

Added: freeswitch/trunk/libs/voipcodecs/src/Makefile.am
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/Makefile.am	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,126 @@
+##
+## VoIPcodecs - a series of DSP components for telephony
+##
+## Makefile.am -- Process this file with automake to produce Makefile.in
+##
+## This program is free software; you can redistribute it and/or modify
+## it under the terms of the GNU General Public License version 2, as
+## published by the Free Software Foundation.
+##
+## This program is distributed in the hope that it will be useful,
+## but WITHOUT ANY WARRANTY; without even the implied warranty of
+## MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+## GNU General Public License for more details.
+##
+## You should have received a copy of the GNU General Public License
+## along with this program; if not, write to the Free Software
+## Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+##
+## $Id: Makefile.am,v 1.85 2008/02/06 09:23:26 steveu Exp $
+
+AM_CFLAGS = $(COMP_VENDOR_CFLAGS)
+
+MAINTAINERCLEANFILES = Makefile.in
+
+INCLUDES = -I$(top_builddir)
+
+lib_LTLIBRARIES = libvoipcodecs.la
+
+libvoipcodecs_la_SOURCES = bitstream.c \
+                           g711.c \
+                           g722_encode.c \
+                           g722_decode.c \
+                           g726.c \
+                           gsm0610_decode.c \
+                           gsm0610_encode.c \
+                           gsm0610_long_term.c \
+                           gsm0610_lpc.c \
+                           gsm0610_preprocess.c \
+                           gsm0610_rpe.c \
+                           gsm0610_short_term.c \
+                           ima_adpcm.c \
+                           lpc10_analyse.c \
+                           lpc10_decode.c \
+                           lpc10_encode.c \
+                           lpc10_placev.c \
+                           lpc10_voicing.c \
+                           oki_adpcm.c \
+                           vector_int.c
+
+libvoipcodecs_la_LDFLAGS = -version-info @VOIPCODECS_LT_CURRENT@:@VOIPCODECS_LT_REVISION@:@VOIPCODECS_LT_AGE@
+
+nobase_include_HEADERS = voipcodecs/bit_operations.h \
+                         voipcodecs/bitstream.h \
+                         voipcodecs/dc_restore.h \
+                         voipcodecs/g711.h \
+                         voipcodecs/g722.h \
+                         voipcodecs/g726.h \
+                         voipcodecs/gsm0610.h \
+                         voipcodecs/ima_adpcm.h \
+                         voipcodecs/lpc10.h \
+                         voipcodecs/oki_adpcm.h \
+                         voipcodecs/telephony.h \
+                         voipcodecs/vector_int.h \
+                         voipcodecs/version.h
+
+nodist_include_HEADERS = voipcodecs.h
+
+noinst_HEADERS = gsm0610_local.h \
+                 lpc10_encdecs.h
+
+# We need to run at_dictionary_gen, so it generates the
+# at_interpreter_dictionary.h file
+
+DSP = libvoipcodecs.dsp
+VCPROJ = libvoipcodecs.vcproj
+
+WIN32SOURCES = $(libvoipcodecs_la_SOURCES) msvc/gettimeofday.c
+WIN32HEADERS = $(nobase_include_HEADERS) voipcodecs.h
+
+DSPOUT = | awk '{printf("%s\r\n", $$0)}' >> $(DSP)
+VCPROJOUT = | awk '{printf("%s\r\n", $$0)}' >> $(VCPROJ)
+
+$(DSP): msvc/msvcproj.head msvc/msvcproj.foot Makefile.am
+	echo "creating $(DSP)"
+	@(cp $(srcdir)/msvc/msvcproj.head $(DSP); \
+	echo "# Begin Group \"Source Files\"" $(DSPOUT); \
+	for file in $(WIN32SOURCES); do \
+	echo "# Begin Source File" $(DSPOUT); \
+	echo "" $(DSPOUT); \
+	echo "SOURCE=.\\"$$file $(DSPOUT); \
+	echo "# End Source File" $(DSPOUT); \
+	done; \
+	echo "# End Group" $(DSPOUT); \
+	echo "# Begin Group \"Header Files\"" $(DSPOUT); \
+	for file in $(WIN32HEADERS); do \
+	echo "# Begin Source File" $(DSPOUT); \
+	echo "" $(DSPOUT); \
+	echo "SOURCE=.\\"$$file $(DSPOUT); \
+	echo "# End Source File" $(DSPOUT); \
+	done; \
+	echo "# End Group" $(DSPOUT); \
+	cat $(srcdir)/msvc/msvcproj.foot $(DSPOUT) )
+
+$(VCPROJ): msvc/vc8proj.head msvc/vc8proj.foot Makefile.am
+	echo "creating $(VCPROJ)"
+	@(cp $(srcdir)/msvc/vc8proj.head $(VCPROJ); \
+	for file in $(WIN32SOURCES); do \
+		echo "<File RelativePath=\""$$file"\"></File>" $(VCPROJOUT); \
+	done; \
+	echo "</Filter><Filter  Name=\"Header Files\">" $(VCPROJOUT); \
+	for file in $(WIN32HEADERS); do \
+		echo "<File RelativePath=\""$$file"\"></File>" $(VCPROJOUT); \
+	done; \
+	cat $(srcdir)/msvc/vc8proj.foot $(VCPROJOUT) )
+
+voipcodecs/version.h:
+	NOWDATE=`date --utc +"%Y%m%d"` ; \
+	NOWTIME=`date --utc +"%H%M%S"` ; \
+	sed 's/$$VOIPCODECS_RELEASE_DATE/'$$NOWDATE'/;s/$$VOIPCODECS_RELEASE_TIME/'$$NOWTIME'/' \
+		<voipcodecs/version.h.in >voipcodecs/version.h
+
+dist-hook:
+	NOWDATE=`date --utc +"%Y%m%d"` ; \
+	NOWTIME=`date --utc +"%H%M%S"` ; \
+	sed 's/$$VOIPCODECS_RELEASE_DATE/'$$NOWDATE'/;s/$$VOIPCODECS_RELEASE_TIME/'$$NOWTIME'/' \
+		<voipcodecs/version.h.in >voipcodecs/version.h

Added: freeswitch/trunk/libs/voipcodecs/src/bitstream.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/bitstream.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,137 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * bitstream.c - Bitstream composition and decomposition routines.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser Lesser GNU General Public License version 2.1.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: bitstream.c,v 1.8 2007/08/20 15:22:21 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <inttypes.h>
+#include <stdlib.h>
+#include <string.h>
+#include <assert.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/bitstream.h"
+
+void bitstream_put(bitstream_state_t *s, uint8_t **c, unsigned int value, int bits)
+{
+    value &= ((1 << bits) - 1);
+    if (s->residue + bits <= 32)
+    {
+        s->bitstream |= (value << s->residue);
+        s->residue += bits;
+    }
+    while (s->residue >= 8)
+    {
+        s->residue -= 8;
+        *(*c)++ = (uint8_t) (s->bitstream & 0xFF);
+        s->bitstream >>= 8;
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+void bitstream_put2(bitstream_state_t *s, uint8_t **c, unsigned int value, int bits)
+{
+    value &= ((1 << bits) - 1);
+    if (s->residue + bits <= 32)
+    {
+        s->bitstream = (s->bitstream << bits) | value;
+        s->residue += bits;
+    }
+    while (s->residue >= 8)
+    {
+        s->residue -= 8;
+        *(*c)++ = (uint8_t) ((s->bitstream >> s->residue) & 0xFF);
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+unsigned int bitstream_get(bitstream_state_t *s, const uint8_t **c, int bits)
+{
+    unsigned int x;
+
+    while (s->residue < (unsigned int) bits)
+    {
+        x = (unsigned int) *(*c)++;
+        s->bitstream |= (x << s->residue);
+        s->residue += 8;
+    }
+    s->residue -= bits;
+    x = s->bitstream & ((1 << bits) - 1);
+    s->bitstream >>= bits;
+    return x;
+}
+/*- End of function --------------------------------------------------------*/
+
+unsigned int bitstream_get2(bitstream_state_t *s, const uint8_t **c, int bits)
+{
+    unsigned int x;
+
+    while (s->residue < (unsigned int) bits)
+    {
+        x = (unsigned int) *(*c)++;
+        s->bitstream = (s->bitstream << 8) | x;
+        s->residue += 8;
+    }
+    s->residue -= bits;
+    x = (s->bitstream >> s->residue) & ((1 << bits) - 1);
+    return x;
+}
+/*- End of function --------------------------------------------------------*/
+
+void bitstream_flush(bitstream_state_t *s, uint8_t **c)
+{
+    if (s->residue > 0)
+    {
+        *(*c)++ = (uint8_t) ((s->bitstream << (8 - s->residue)) & 0xFF);
+        s->residue = 0;
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+void bitstream_flush2(bitstream_state_t *s, uint8_t **c)
+{
+    if (s->residue > 0)
+    {
+        *(*c)++ = (uint8_t) ((s->bitstream << (8 - s->residue)) & 0xFF);
+        s->residue = 0;
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+bitstream_state_t *bitstream_init(bitstream_state_t *s)
+{
+    if (s == NULL)
+        return NULL;
+    s->bitstream = 0;
+    s->residue = 0;
+    return s;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/float_fudge.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/float_fudge.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,123 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * float_fudge.h - A bunch of shims, to use double maths
+ *                 functions on platforms which lack the
+ *                 float versions with an 'f' at the end.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2004 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: float_fudge.h,v 1.2 2007/08/13 11:35:32 steveu Exp $
+ */
+
+#if !defined(_FLOAT_FUDGE_H_)
+#define _FLOAT_FUDGE_H_
+
+#if defined(__USE_DOUBLE_MATH__)
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+static __inline__ float sinf(float x)
+{
+	return (float) sin((double) x);
+}
+
+static __inline__ float cosf(float x)
+{
+	return (float) cos((double) x);
+}
+
+static __inline__ float tanf(float x)
+{
+	return (float) tan((double) x);
+}
+
+static __inline__ float asinf(float x)
+{
+	return (float) asin((double) x);
+}
+
+static __inline__ float acosf(float x)
+{
+	return (float) acos((double) x);
+}
+
+static __inline__ float atanf(float x)
+{
+	return (float) atan((double) x);
+}
+
+static __inline__ float atan2f(float y, float x)
+{
+	return (float) atan2((double) y, (double) x);
+}
+
+static __inline__ float ceilf(float x)
+{
+	return (float) ceil((double) x);
+}
+
+static __inline__ float floorf(float x)
+{
+	return (float) floor((double) x);
+}
+
+static __inline__ float expf(float x)
+{
+    return (float) expf((double) x);
+}
+
+static __inline__ float logf(float x)
+{
+	return (float) logf((double) x);
+}
+
+static __inline__ float log10f(float x)
+{
+    return (float) log10((double) x);
+}
+
+static __inline__ float powf(float x, float y)
+{
+    return (float) pow((double) x, (double) y);
+}
+
+static __inline__ int rintf(float x)
+{
+	return (int) rint((double) x);
+}
+
+static __inline__ long int lrintf(float x)
+{
+    return (long int) lrint((double) x);
+}
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+
+#endif
+
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/g711.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/g711.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,104 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * g711.c - A-law and u-law transcoding routines
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser Lesser GNU General Public License version 2.1.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: g711.c,v 1.4 2006/11/19 14:07:24 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <inttypes.h>
+#include <stdlib.h>
+#include <string.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+#include <assert.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/bit_operations.h"
+#include "voipcodecs/g711.h"
+
+/* Copied from the CCITT G.711 specification */
+static const uint8_t ulaw_to_alaw_table[256] =
+{
+     42,  43,  40,  41,  46,  47,  44,  45,  34,  35,  32,  33,  38,  39,  36,  37,
+     58,  59,  56,  57,  62,  63,  60,  61,  50,  51,  48,  49,  54,  55,  52,  53,
+     10,  11,   8,   9,  14,  15,  12,  13,   2,   3,   0,   1,   6,   7,   4,  26,
+     27,  24,  25,  30,  31,  28,  29,  18,  19,  16,  17,  22,  23,  20,  21, 106,
+    104, 105, 110, 111, 108, 109,  98,  99,  96,  97, 102, 103, 100, 101, 122, 120,
+    126, 127, 124, 125, 114, 115, 112, 113, 118, 119, 116, 117,  75,  73,  79,  77,
+     66,  67,  64,  65,  70,  71,  68,  69,  90,  91,  88,  89,  94,  95,  92,  93,
+     82,  82,  83,  83,  80,  80,  81,  81,  86,  86,  87,  87,  84,  84,  85,  85,
+    170, 171, 168, 169, 174, 175, 172, 173, 162, 163, 160, 161, 166, 167, 164, 165,
+    186, 187, 184, 185, 190, 191, 188, 189, 178, 179, 176, 177, 182, 183, 180, 181,
+    138, 139, 136, 137, 142, 143, 140, 141, 130, 131, 128, 129, 134, 135, 132, 154,
+    155, 152, 153, 158, 159, 156, 157, 146, 147, 144, 145, 150, 151, 148, 149, 234,
+    232, 233, 238, 239, 236, 237, 226, 227, 224, 225, 230, 231, 228, 229, 250, 248,
+    254, 255, 252, 253, 242, 243, 240, 241, 246, 247, 244, 245, 203, 201, 207, 205,
+    194, 195, 192, 193, 198, 199, 196, 197, 218, 219, 216, 217, 222, 223, 220, 221,
+    210, 210, 211, 211, 208, 208, 209, 209, 214, 214, 215, 215, 212, 212, 213, 213
+};
+
+/* These transcoding tables are copied from the CCITT G.711 specification. To achieve
+   optimal results, do not change them. */
+
+static const uint8_t alaw_to_ulaw_table[256] =
+{
+     42,  43,  40,  41,  46,  47,  44,  45,  34,  35,  32,  33,  38,  39,  36,  37,
+     57,  58,  55,  56,  61,  62,  59,  60,  49,  50,  47,  48,  53,  54,  51,  52,
+     10,  11,   8,   9,  14,  15,  12,  13,   2,   3,   0,   1,   6,   7,   4,   5,
+     26,  27,  24,  25,  30,  31,  28,  29,  18,  19,  16,  17,  22,  23,  20,  21,
+     98,  99,  96,  97, 102, 103, 100, 101,  93,  93,  92,  92,  95,  95,  94,  94,
+    116, 118, 112, 114, 124, 126, 120, 122, 106, 107, 104, 105, 110, 111, 108, 109,
+     72,  73,  70,  71,  76,  77,  74,  75,  64,  65,  63,  63,  68,  69,  66,  67,
+     86,  87,  84,  85,  90,  91,  88,  89,  79,  79,  78,  78,  82,  83,  80,  81,
+    170, 171, 168, 169, 174, 175, 172, 173, 162, 163, 160, 161, 166, 167, 164, 165,
+    185, 186, 183, 184, 189, 190, 187, 188, 177, 178, 175, 176, 181, 182, 179, 180,
+    138, 139, 136, 137, 142, 143, 140, 141, 130, 131, 128, 129, 134, 135, 132, 133,
+    154, 155, 152, 153, 158, 159, 156, 157, 146, 147, 144, 145, 150, 151, 148, 149,
+    226, 227, 224, 225, 230, 231, 228, 229, 221, 221, 220, 220, 223, 223, 222, 222,
+    244, 246, 240, 242, 252, 254, 248, 250, 234, 235, 232, 233, 238, 239, 236, 237,
+    200, 201, 198, 199, 204, 205, 202, 203, 192, 193, 191, 191, 196, 197, 194, 195,
+    214, 215, 212, 213, 218, 219, 216, 217, 207, 207, 206, 206, 210, 211, 208, 209
+};
+
+uint8_t alaw_to_ulaw(uint8_t alaw)
+{
+    return alaw_to_ulaw_table[alaw];
+}
+/*- End of function --------------------------------------------------------*/
+
+uint8_t ulaw_to_alaw(uint8_t ulaw)
+{
+    return ulaw_to_alaw_table[ulaw];
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/g722_decode.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/g722_decode.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,404 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * g722_decode.c - The ITU G.722 codec, decode part.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser Lesser GNU General Public License version 2.1.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * Based in part on a single channel G.722 codec which is:
+ *
+ * Copyright (c) CMU 1993
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722_decode.c,v 1.19 2007/03/14 11:51:01 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <inttypes.h>
+#include <memory.h>
+#include <stdlib.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/g722.h"
+
+static void block4(g722_decode_state_t *s, int band, int d);
+
+static void block4(g722_decode_state_t *s, int band, int d)
+{
+    int wd1;
+    int wd2;
+    int wd3;
+    int i;
+
+    /* Block 4, RECONS */
+    s->band[band].d[0] = d;
+    s->band[band].r[0] = saturate(s->band[band].s + d);
+
+    /* Block 4, PARREC */
+    s->band[band].p[0] = saturate(s->band[band].sz + d);
+
+    /* Block 4, UPPOL2 */
+    for (i = 0;  i < 3;  i++)
+        s->band[band].sg[i] = s->band[band].p[i] >> 15;
+    wd1 = saturate(s->band[band].a[1] << 2);
+
+    wd2 = (s->band[band].sg[0] == s->band[band].sg[1])  ?  -wd1  :  wd1;
+    if (wd2 > 32767)
+        wd2 = 32767;
+    wd3 = (s->band[band].sg[0] == s->band[band].sg[2])  ?  128  :  -128;
+    wd3 += (wd2 >> 7);
+    wd3 += (s->band[band].a[2]*32512) >> 15;
+    if (wd3 > 12288)
+        wd3 = 12288;
+    else if (wd3 < -12288)
+        wd3 = -12288;
+    s->band[band].ap[2] = wd3;
+
+    /* Block 4, UPPOL1 */
+    s->band[band].sg[0] = s->band[band].p[0] >> 15;
+    s->band[band].sg[1] = s->band[band].p[1] >> 15;
+    wd1 = (s->band[band].sg[0] == s->band[band].sg[1])  ?  192  :  -192;
+    wd2 = (s->band[band].a[1]*32640) >> 15;
+
+    s->band[band].ap[1] = saturate(wd1 + wd2);
+    wd3 = saturate(15360 - s->band[band].ap[2]);
+    if (s->band[band].ap[1] > wd3)
+        s->band[band].ap[1] = wd3;
+    else if (s->band[band].ap[1] < -wd3)
+        s->band[band].ap[1] = -wd3;
+
+    /* Block 4, UPZERO */
+    wd1 = (d == 0)  ?  0  :  128;
+    s->band[band].sg[0] = d >> 15;
+    for (i = 1;  i < 7;  i++)
+    {
+        s->band[band].sg[i] = s->band[band].d[i] >> 15;
+        wd2 = (s->band[band].sg[i] == s->band[band].sg[0])  ?  wd1  :  -wd1;
+        wd3 = (s->band[band].b[i]*32640) >> 15;
+        s->band[band].bp[i] = saturate(wd2 + wd3);
+    }
+
+    /* Block 4, DELAYA */
+    for (i = 6;  i > 0;  i--)
+    {
+        s->band[band].d[i] = s->band[band].d[i - 1];
+        s->band[band].b[i] = s->band[band].bp[i];
+    }
+    
+    for (i = 2;  i > 0;  i--)
+    {
+        s->band[band].r[i] = s->band[band].r[i - 1];
+        s->band[band].p[i] = s->band[band].p[i - 1];
+        s->band[band].a[i] = s->band[band].ap[i];
+    }
+
+    /* Block 4, FILTEP */
+    wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
+    wd1 = (s->band[band].a[1]*wd1) >> 15;
+    wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
+    wd2 = (s->band[band].a[2]*wd2) >> 15;
+    s->band[band].sp = saturate(wd1 + wd2);
+
+    /* Block 4, FILTEZ */
+    s->band[band].sz = 0;
+    for (i = 6;  i > 0;  i--)
+    {
+        wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
+        s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
+    }
+    s->band[band].sz = saturate(s->band[band].sz);
+
+    /* Block 4, PREDIC */
+    s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
+}
+/*- End of function --------------------------------------------------------*/
+
+g722_decode_state_t *g722_decode_init(g722_decode_state_t *s, int rate, int options)
+{
+    if (s == NULL)
+    {
+        if ((s = (g722_decode_state_t *) malloc(sizeof(*s))) == NULL)
+            return NULL;
+    }
+    memset(s, 0, sizeof(*s));
+    if (rate == 48000)
+        s->bits_per_sample = 6;
+    else if (rate == 56000)
+        s->bits_per_sample = 7;
+    else
+        s->bits_per_sample = 8;
+    if ((options & G722_SAMPLE_RATE_8000))
+        s->eight_k = TRUE;
+    if ((options & G722_PACKED)  &&  s->bits_per_sample != 8)
+        s->packed = TRUE;
+    else
+        s->packed = FALSE;
+    s->band[0].det = 32;
+    s->band[1].det = 8;
+    return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int g722_decode_release(g722_decode_state_t *s)
+{
+    free(s);
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[], int len)
+{
+    static const int wl[8] =
+    {
+        -60, -30, 58, 172, 334, 538, 1198, 3042
+    };
+    static const int rl42[16] =
+    {
+        0, 7, 6, 5, 4, 3, 2, 1, 7, 6, 5, 4, 3,  2, 1, 0
+    };
+    static const int ilb[32] =
+    {
+        2048, 2093, 2139, 2186, 2233, 2282, 2332,
+        2383, 2435, 2489, 2543, 2599, 2656, 2714,
+        2774, 2834, 2896, 2960, 3025, 3091, 3158,
+        3228, 3298, 3371, 3444, 3520, 3597, 3676,
+        3756, 3838, 3922, 4008
+    };
+    static const int wh[3] =
+    {
+        0, -214, 798
+    };
+    static const int rh2[4] =
+    {
+        2, 1, 2, 1
+    };
+    static const int qm2[4] =
+    {
+        -7408, -1616,  7408,   1616
+    };
+    static const int qm4[16] = 
+    {
+              0, -20456, -12896,  -8968, 
+          -6288,  -4240,  -2584,  -1200,
+          20456,  12896,   8968,   6288,
+           4240,   2584,   1200,      0
+    };
+    static const int qm5[32] =
+    {
+           -280,   -280, -23352, -17560,
+         -14120, -11664,  -9752,  -8184,
+          -6864,  -5712,  -4696,  -3784,
+          -2960,  -2208,  -1520,   -880,
+          23352,  17560,  14120,  11664,
+           9752,   8184,   6864,   5712,
+           4696,   3784,   2960,   2208,
+           1520,    880,    280,   -280
+    };
+    static const int qm6[64] =
+    {
+           -136,   -136,   -136,   -136,
+         -24808, -21904, -19008, -16704,
+         -14984, -13512, -12280, -11192,
+         -10232,  -9360,  -8576,  -7856,
+          -7192,  -6576,  -6000,  -5456,
+          -4944,  -4464,  -4008,  -3576,
+          -3168,  -2776,  -2400,  -2032,
+          -1688,  -1360,  -1040,   -728,
+          24808,  21904,  19008,  16704,
+          14984,  13512,  12280,  11192,
+          10232,   9360,   8576,   7856,
+           7192,   6576,   6000,   5456,
+           4944,   4464,   4008,   3576,
+           3168,   2776,   2400,   2032,
+           1688,   1360,   1040,    728,
+            432,    136,   -432,   -136
+    };
+    static const int qmf_coeffs[12] =
+    {
+           3,  -11,   12,   32, -210,  951, 3876, -805,  362, -156,   53,  -11,
+    };
+    int dlowt;
+    int rlow;
+    int ihigh;
+    int dhigh;
+    int rhigh;
+    int xout1;
+    int xout2;
+    int wd1;
+    int wd2;
+    int wd3;
+    int code;
+    int outlen;
+    int i;
+    int j;
+
+    outlen = 0;
+    rhigh = 0;
+    for (j = 0;  j < len;  )
+    {
+        if (s->packed)
+        {
+            /* Unpack the code bits */
+            if (s->in_bits < s->bits_per_sample)
+            {
+                s->in_buffer |= (g722_data[j++] << s->in_bits);
+                s->in_bits += 8;
+            }
+            code = s->in_buffer & ((1 << s->bits_per_sample) - 1);
+            s->in_buffer >>= s->bits_per_sample;
+            s->in_bits -= s->bits_per_sample;
+        }
+        else
+        {
+            code = g722_data[j++];
+        }
+
+        switch (s->bits_per_sample)
+        {
+        default:
+        case 8:
+            wd1 = code & 0x3F;
+            ihigh = (code >> 6) & 0x03;
+            wd2 = qm6[wd1];
+            wd1 >>= 2;
+            break;
+        case 7:
+            wd1 = code & 0x1F;
+            ihigh = (code >> 5) & 0x03;
+            wd2 = qm5[wd1];
+            wd1 >>= 1;
+            break;
+        case 6:
+            wd1 = code & 0x0F;
+            ihigh = (code >> 4) & 0x03;
+            wd2 = qm4[wd1];
+            break;
+        }
+        /* Block 5L, LOW BAND INVQBL */
+        wd2 = (s->band[0].det*wd2) >> 15;
+        /* Block 5L, RECONS */
+        rlow = s->band[0].s + wd2;
+        /* Block 6L, LIMIT */
+        if (rlow > 16383)
+            rlow = 16383;
+        else if (rlow < -16384)
+            rlow = -16384;
+
+        /* Block 2L, INVQAL */
+        wd2 = qm4[wd1];
+        dlowt = (s->band[0].det*wd2) >> 15;
+
+        /* Block 3L, LOGSCL */
+        wd2 = rl42[wd1];
+        wd1 = (s->band[0].nb*127) >> 7;
+        wd1 += wl[wd2];
+        if (wd1 < 0)
+            wd1 = 0;
+        else if (wd1 > 18432)
+            wd1 = 18432;
+        s->band[0].nb = wd1;
+            
+        /* Block 3L, SCALEL */
+        wd1 = (s->band[0].nb >> 6) & 31;
+        wd2 = 8 - (s->band[0].nb >> 11);
+        wd3 = (wd2 < 0)  ?  (ilb[wd1] << -wd2)  :  (ilb[wd1] >> wd2);
+        s->band[0].det = wd3 << 2;
+
+        block4(s, 0, dlowt);
+        
+        if (!s->eight_k)
+        {
+            /* Block 2H, INVQAH */
+            wd2 = qm2[ihigh];
+            dhigh = (s->band[1].det*wd2) >> 15;
+            /* Block 5H, RECONS */
+            rhigh = dhigh + s->band[1].s;
+            /* Block 6H, LIMIT */
+            if (rhigh > 16383)
+                rhigh = 16383;
+            else if (rhigh < -16384)
+                rhigh = -16384;
+
+            /* Block 2H, INVQAH */
+            wd2 = rh2[ihigh];
+            wd1 = (s->band[1].nb*127) >> 7;
+            wd1 += wh[wd2];
+            if (wd1 < 0)
+                wd1 = 0;
+            else if (wd1 > 22528)
+                wd1 = 22528;
+            s->band[1].nb = wd1;
+            
+            /* Block 3H, SCALEH */
+            wd1 = (s->band[1].nb >> 6) & 31;
+            wd2 = 10 - (s->band[1].nb >> 11);
+            wd3 = (wd2 < 0)  ?  (ilb[wd1] << -wd2)  :  (ilb[wd1] >> wd2);
+            s->band[1].det = wd3 << 2;
+
+            block4(s, 1, dhigh);
+        }
+
+        if (s->itu_test_mode)
+        {
+            amp[outlen++] = (int16_t) (rlow << 1);
+            amp[outlen++] = (int16_t) (rhigh << 1);
+        }
+        else
+        {
+            if (s->eight_k)
+            {
+                amp[outlen++] = (int16_t) rlow;
+            }
+            else
+            {
+                /* Apply the receive QMF */
+                memcpy(s->x, &s->x[2], 22*sizeof(s->x[0]));
+                s->x[22] = rlow + rhigh;
+                s->x[23] = rlow - rhigh;
+
+                xout1 = 0;
+                xout2 = 0;
+                for (i = 0;  i < 12;  i++)
+                {
+                    xout2 += s->x[2*i]*qmf_coeffs[i];
+                    xout1 += s->x[2*i + 1]*qmf_coeffs[11 - i];
+                }
+                amp[outlen++] = (int16_t) (xout1 >> 12);
+                amp[outlen++] = (int16_t) (xout2 >> 12);
+            }
+        }
+    }
+    return outlen;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/g722_encode.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/g722_encode.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,390 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * g722_encode.c - The ITU G.722 codec, encode part.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser Lesser GNU General Public License version 2.1.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * Based on a single channel 64kbps only G.722 codec which is:
+ *
+ *****    Copyright (c) CMU    1993      *****
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722_encode.c,v 1.17 2007/03/14 11:51:01 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <inttypes.h>
+#include <memory.h>
+#include <stdlib.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/g722.h"
+
+static void block4(g722_encode_state_t *s, int band, int d)
+{
+    int wd1;
+    int wd2;
+    int wd3;
+    int i;
+
+    /* Block 4, RECONS */
+    s->band[band].d[0] = d;
+    s->band[band].r[0] = saturate(s->band[band].s + d);
+
+    /* Block 4, PARREC */
+    s->band[band].p[0] = saturate(s->band[band].sz + d);
+
+    /* Block 4, UPPOL2 */
+    for (i = 0;  i < 3;  i++)
+        s->band[band].sg[i] = s->band[band].p[i] >> 15;
+    wd1 = saturate(s->band[band].a[1] << 2);
+
+    wd2 = (s->band[band].sg[0] == s->band[band].sg[1])  ?  -wd1  :  wd1;
+    if (wd2 > 32767)
+        wd2 = 32767;
+    wd3 = (wd2 >> 7) + ((s->band[band].sg[0] == s->band[band].sg[2])  ?  128  :  -128);
+    wd3 += (s->band[band].a[2]*32512) >> 15;
+    if (wd3 > 12288)
+        wd3 = 12288;
+    else if (wd3 < -12288)
+        wd3 = -12288;
+    s->band[band].ap[2] = wd3;
+
+    /* Block 4, UPPOL1 */
+    s->band[band].sg[0] = s->band[band].p[0] >> 15;
+    s->band[band].sg[1] = s->band[band].p[1] >> 15;
+    wd1 = (s->band[band].sg[0] == s->band[band].sg[1])  ?  192  :  -192;
+    wd2 = (s->band[band].a[1]*32640) >> 15;
+
+    s->band[band].ap[1] = saturate(wd1 + wd2);
+    wd3 = saturate(15360 - s->band[band].ap[2]);
+    if (s->band[band].ap[1] > wd3)
+        s->band[band].ap[1] = wd3;
+    else if (s->band[band].ap[1] < -wd3)
+        s->band[band].ap[1] = -wd3;
+
+    /* Block 4, UPZERO */
+    wd1 = (d == 0)  ?  0  :  128;
+    s->band[band].sg[0] = d >> 15;
+    for (i = 1;  i < 7;  i++)
+    {
+        s->band[band].sg[i] = s->band[band].d[i] >> 15;
+        wd2 = (s->band[band].sg[i] == s->band[band].sg[0])  ?  wd1  :  -wd1;
+        wd3 = (s->band[band].b[i]*32640) >> 15;
+        s->band[band].bp[i] = saturate(wd2 + wd3);
+    }
+
+    /* Block 4, DELAYA */
+    for (i = 6;  i > 0;  i--)
+    {
+        s->band[band].d[i] = s->band[band].d[i - 1];
+        s->band[band].b[i] = s->band[band].bp[i];
+    }
+    
+    for (i = 2;  i > 0;  i--)
+    {
+        s->band[band].r[i] = s->band[band].r[i - 1];
+        s->band[band].p[i] = s->band[band].p[i - 1];
+        s->band[band].a[i] = s->band[band].ap[i];
+    }
+
+    /* Block 4, FILTEP */
+    wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
+    wd1 = (s->band[band].a[1]*wd1) >> 15;
+    wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
+    wd2 = (s->band[band].a[2]*wd2) >> 15;
+    s->band[band].sp = saturate(wd1 + wd2);
+
+    /* Block 4, FILTEZ */
+    s->band[band].sz = 0;
+    for (i = 6;  i > 0;  i--)
+    {
+        wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
+        s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
+    }
+    s->band[band].sz = saturate(s->band[band].sz);
+
+    /* Block 4, PREDIC */
+    s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
+}
+/*- End of function --------------------------------------------------------*/
+
+g722_encode_state_t *g722_encode_init(g722_encode_state_t *s, int rate, int options)
+{
+    if (s == NULL)
+    {
+        if ((s = (g722_encode_state_t *) malloc(sizeof(*s))) == NULL)
+            return NULL;
+    }
+    memset(s, 0, sizeof(*s));
+    if (rate == 48000)
+        s->bits_per_sample = 6;
+    else if (rate == 56000)
+        s->bits_per_sample = 7;
+    else
+        s->bits_per_sample = 8;
+    if ((options & G722_SAMPLE_RATE_8000))
+        s->eight_k = TRUE;
+    if ((options & G722_PACKED)  &&  s->bits_per_sample != 8)
+        s->packed = TRUE;
+    else
+        s->packed = FALSE;
+    s->band[0].det = 32;
+    s->band[1].det = 8;
+    return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int g722_encode_release(g722_encode_state_t *s)
+{
+    free(s);
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+int g722_encode(g722_encode_state_t *s, uint8_t g722_data[], const int16_t amp[], int len)
+{
+    static const int q6[32] =
+    {
+           0,   35,   72,  110,  150,  190,  233,  276,
+         323,  370,  422,  473,  530,  587,  650,  714,
+         786,  858,  940, 1023, 1121, 1219, 1339, 1458,
+        1612, 1765, 1980, 2195, 2557, 2919,    0,    0
+    };
+    static const int iln[32] =
+    {
+         0, 63, 62, 31, 30, 29, 28, 27,
+        26, 25, 24, 23, 22, 21, 20, 19,
+        18, 17, 16, 15, 14, 13, 12, 11,
+        10,  9,  8,  7,  6,  5,  4,  0
+    };
+    static const int ilp[32] =
+    {
+         0, 61, 60, 59, 58, 57, 56, 55,
+        54, 53, 52, 51, 50, 49, 48, 47,
+        46, 45, 44, 43, 42, 41, 40, 39,
+        38, 37, 36, 35, 34, 33, 32,  0
+    };
+    static const int wl[8] =
+    {
+        -60, -30, 58, 172, 334, 538, 1198, 3042
+    };
+    static const int rl42[16] =
+    {
+        0, 7, 6, 5, 4, 3, 2, 1, 7, 6, 5, 4, 3, 2, 1, 0
+    };
+    static const int ilb[32] =
+    {
+        2048, 2093, 2139, 2186, 2233, 2282, 2332,
+        2383, 2435, 2489, 2543, 2599, 2656, 2714,
+        2774, 2834, 2896, 2960, 3025, 3091, 3158,
+        3228, 3298, 3371, 3444, 3520, 3597, 3676,
+        3756, 3838, 3922, 4008
+    };
+    static const int qm4[16] =
+    {
+             0, -20456, -12896, -8968,
+         -6288,  -4240,  -2584, -1200,
+         20456,  12896,   8968,  6288,
+          4240,   2584,   1200,     0
+    };
+    static const int qm2[4] =
+    {
+        -7408,  -1616,   7408,   1616
+    };
+    static const int qmf_coeffs[12] =
+    {
+           3,  -11,   12,   32, -210,  951, 3876, -805,  362, -156,   53,  -11,
+    };
+    static const int ihn[3] = {0, 1, 0};
+    static const int ihp[3] = {0, 3, 2};
+    static const int wh[3] = {0, -214, 798};
+    static const int rh2[4] = {2, 1, 2, 1};
+
+    int dlow;
+    int dhigh;
+    int el;
+    int wd;
+    int wd1;
+    int ril;
+    int wd2;
+    int il4;
+    int ih2;
+    int wd3;
+    int eh;
+    int mih;
+    int i;
+    int j;
+    /* Low and high band PCM from the QMF */
+    int xlow;
+    int xhigh;
+    int g722_bytes;
+    /* Even and odd tap accumulators */
+    int sumeven;
+    int sumodd;
+    int ihigh;
+    int ilow;
+    int code;
+
+    g722_bytes = 0;
+    xhigh = 0;
+    for (j = 0;  j < len;  )
+    {
+        if (s->itu_test_mode)
+        {
+            xlow =
+            xhigh = amp[j++] >> 1;
+        }
+        else
+        {
+            if (s->eight_k)
+            {
+                xlow = amp[j++];
+            }
+            else
+            {
+                /* Apply the transmit QMF */
+                /* Shuffle the buffer down */
+                memcpy(s->x, &s->x[2], 22*sizeof(s->x[0]));
+                s->x[22] = amp[j++];
+                s->x[23] = amp[j++];
+    
+                /* Discard every other QMF output */
+                sumeven = 0;
+                sumodd = 0;
+                for (i = 0;  i < 12;  i++)
+                {
+                    sumodd += s->x[2*i]*qmf_coeffs[i];
+                    sumeven += s->x[2*i + 1]*qmf_coeffs[11 - i];
+                }
+                xlow = (sumeven + sumodd) >> 13;
+                xhigh = (sumeven - sumodd) >> 13;
+            }
+        }
+        /* Block 1L, SUBTRA */
+        el = saturate(xlow - s->band[0].s);
+
+        /* Block 1L, QUANTL */
+        wd = (el >= 0)  ?  el  :  -(el + 1);
+
+        for (i = 1;  i < 30;  i++)
+        {
+            wd1 = (q6[i]*s->band[0].det) >> 12;
+            if (wd < wd1)
+                break;
+        }
+        ilow = (el < 0)  ?  iln[i]  :  ilp[i];
+
+        /* Block 2L, INVQAL */
+        ril = ilow >> 2;
+        wd2 = qm4[ril];
+        dlow = (s->band[0].det*wd2) >> 15;
+
+        /* Block 3L, LOGSCL */
+        il4 = rl42[ril];
+        wd = (s->band[0].nb*127) >> 7;
+        s->band[0].nb = wd + wl[il4];
+        if (s->band[0].nb < 0)
+            s->band[0].nb = 0;
+        else if (s->band[0].nb > 18432)
+            s->band[0].nb = 18432;
+
+        /* Block 3L, SCALEL */
+        wd1 = (s->band[0].nb >> 6) & 31;
+        wd2 = 8 - (s->band[0].nb >> 11);
+        wd3 = (wd2 < 0)  ?  (ilb[wd1] << -wd2)  :  (ilb[wd1] >> wd2);
+        s->band[0].det = wd3 << 2;
+
+        block4(s, 0, dlow);
+        
+        if (s->eight_k)
+        {
+            /* Just leave the high bits as zero */
+            code = (0xC0 | ilow) >> (8 - s->bits_per_sample);
+        }
+        else
+        {
+            /* Block 1H, SUBTRA */
+            eh = saturate(xhigh - s->band[1].s);
+
+            /* Block 1H, QUANTH */
+            wd = (eh >= 0)  ?  eh  :  -(eh + 1);
+            wd1 = (564*s->band[1].det) >> 12;
+            mih = (wd >= wd1)  ?  2  :  1;
+            ihigh = (eh < 0)  ?  ihn[mih]  :  ihp[mih];
+
+            /* Block 2H, INVQAH */
+            wd2 = qm2[ihigh];
+            dhigh = (s->band[1].det*wd2) >> 15;
+
+            /* Block 3H, LOGSCH */
+            ih2 = rh2[ihigh];
+            wd = (s->band[1].nb*127) >> 7;
+            s->band[1].nb = wd + wh[ih2];
+            if (s->band[1].nb < 0)
+                s->band[1].nb = 0;
+            else if (s->band[1].nb > 22528)
+                s->band[1].nb = 22528;
+
+            /* Block 3H, SCALEH */
+            wd1 = (s->band[1].nb >> 6) & 31;
+            wd2 = 10 - (s->band[1].nb >> 11);
+            wd3 = (wd2 < 0)  ?  (ilb[wd1] << -wd2)  :  (ilb[wd1] >> wd2);
+            s->band[1].det = wd3 << 2;
+
+            block4(s, 1, dhigh);
+            code = ((ihigh << 6) | ilow) >> (8 - s->bits_per_sample);
+        }
+
+        if (s->packed)
+        {
+            /* Pack the code bits */
+            s->out_buffer |= (code << s->out_bits);
+            s->out_bits += s->bits_per_sample;
+            if (s->out_bits >= 8)
+            {
+                g722_data[g722_bytes++] = (uint8_t) (s->out_buffer & 0xFF);
+                s->out_bits -= 8;
+                s->out_buffer >>= 8;
+            }
+        }
+        else
+        {
+            g722_data[g722_bytes++] = (uint8_t) code;
+        }
+    }
+    return g722_bytes;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/g726.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/g726.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,1178 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * g726.c - The ITU G.726 codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * Based on G.721/G.723 code which is:
+ *
+ * This source code is a product of Sun Microsystems, Inc. and is provided
+ * for unrestricted use.  Users may copy or modify this source code without
+ * charge.
+ *
+ * SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
+ * THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
+ * PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
+ *
+ * Sun source code is provided with no support and without any obligation on
+ * the part of Sun Microsystems, Inc. to assist in its use, correction,
+ * modification or enhancement.
+ *
+ * SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
+ * INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
+ * OR ANY PART THEREOF.
+ *
+ * In no event will Sun Microsystems, Inc. be liable for any lost revenue
+ * or profits or other special, indirect and consequential damages, even if
+ * Sun has been advised of the possibility of such damages.
+ *
+ * Sun Microsystems, Inc.
+ * 2550 Garcia Avenue
+ * Mountain View, California  94043
+ *
+ * $Id: g726.c,v 1.17 2006/11/19 14:07:24 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <inttypes.h>
+#include <memory.h>
+#include <stdlib.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/bitstream.h"
+#include "voipcodecs/bit_operations.h"
+#include "voipcodecs/g711.h"
+#include "voipcodecs/g726.h"
+
+/*
+ * Maps G.726_16 code word to reconstructed scale factor normalized log
+ * magnitude values.
+ */
+static const int g726_16_dqlntab[4] =
+{
+    116, 365, 365, 116
+};
+
+/* Maps G.726_16 code word to log of scale factor multiplier. */
+static const int g726_16_witab[4] =
+{
+    -704, 14048, 14048, -704
+};
+
+/*
+ * Maps G.726_16 code words to a set of values whose long and short
+ * term averages are computed and then compared to give an indication
+ * how stationary (steady state) the signal is.
+ */
+static const int g726_16_fitab[4] =
+{
+    0x000, 0xE00, 0xE00, 0x000
+};
+
+static const int qtab_726_16[1] =
+{
+    261
+};
+
+/*
+ * Maps G.726_24 code word to reconstructed scale factor normalized log
+ * magnitude values.
+ */
+static const int g726_24_dqlntab[8] =
+{
+    -2048, 135, 273, 373, 373, 273, 135, -2048
+};
+
+/* Maps G.726_24 code word to log of scale factor multiplier. */
+static const int g726_24_witab[8] =
+{
+    -128, 960, 4384, 18624, 18624, 4384, 960, -128
+};
+
+/*
+ * Maps G.726_24 code words to a set of values whose long and short
+ * term averages are computed and then compared to give an indication
+ * how stationary (steady state) the signal is.
+ */
+static const int g726_24_fitab[8] =
+{
+    0x000, 0x200, 0x400, 0xE00, 0xE00, 0x400, 0x200, 0x000
+};
+
+static const int qtab_726_24[3] =
+{
+    8, 218, 331
+};
+
+/*
+ * Maps G.726_32 code word to reconstructed scale factor normalized log
+ * magnitude values.
+ */
+static const int g726_32_dqlntab[16] =
+{
+    -2048,   4, 135, 213, 273, 323, 373,   425,
+      425, 373, 323, 273, 213, 135,   4, -2048
+};
+
+/* Maps G.726_32 code word to log of scale factor multiplier. */
+static const int g726_32_witab[16] =
+{
+     -384,   576,  1312,  2048,  3584,  6336, 11360, 35904,
+    35904, 11360,  6336,  3584,  2048,  1312,   576,  -384
+};
+
+/*
+ * Maps G.726_32 code words to a set of values whose long and short
+ * term averages are computed and then compared to give an indication
+ * how stationary (steady state) the signal is.
+ */
+static const int g726_32_fitab[16] =
+{
+    0x000, 0x000, 0x000, 0x200, 0x200, 0x200, 0x600, 0xE00,
+    0xE00, 0x600, 0x200, 0x200, 0x200, 0x000, 0x000, 0x000
+};
+
+static const int qtab_726_32[7] =
+{
+    -124, 80, 178, 246, 300, 349, 400
+};
+
+/*
+ * Maps G.726_40 code word to ructeconstructed scale factor normalized log
+ * magnitude values.
+ */
+static const int g726_40_dqlntab[32] =
+{
+    -2048, -66, 28, 104, 169, 224, 274, 318,
+      358, 395, 429, 459, 488, 514, 539, 566,
+      566, 539, 514, 488, 459, 429, 395, 358,
+      318, 274, 224, 169, 104, 28, -66, -2048
+};
+
+/* Maps G.726_40 code word to log of scale factor multiplier. */
+static const int g726_40_witab[32] =
+{
+      448,   448,   768,  1248,  1280,  1312,  1856,  3200,
+     4512,  5728,  7008,  8960, 11456, 14080, 16928, 22272,
+    22272, 16928, 14080, 11456,  8960,  7008,  5728,  4512,
+     3200,  1856,  1312,  1280,  1248,   768,   448,   448
+};
+
+/*
+ * Maps G.726_40 code words to a set of values whose long and short
+ * term averages are computed and then compared to give an indication
+ * how stationary (steady state) the signal is.
+ */
+static const int g726_40_fitab[32] =
+{
+    0x000, 0x000, 0x000, 0x000, 0x000, 0x200, 0x200, 0x200,
+    0x200, 0x200, 0x400, 0x600, 0x800, 0xA00, 0xC00, 0xC00,
+    0xC00, 0xC00, 0xA00, 0x800, 0x600, 0x400, 0x200, 0x200,
+    0x200, 0x200, 0x200, 0x000, 0x000, 0x000, 0x000, 0x000
+};
+
+static const int qtab_726_40[15] =
+{
+    -122, -16,  68, 139, 198, 250, 298, 339,
+     378, 413, 445, 475, 502, 528, 553
+};
+
+/*
+ * returns the integer product of the 14-bit integer "an" and
+ * "floating point" representation (4-bit exponent, 6-bit mantessa) "srn".
+ */
+static int16_t fmult(int16_t an, int16_t srn)
+{
+    int16_t anmag;
+    int16_t anexp;
+    int16_t anmant;
+    int16_t wanexp;
+    int16_t wanmant;
+    int16_t retval;
+
+    anmag = (an > 0)  ?  an  :  ((-an) & 0x1FFF);
+    anexp = (int16_t) (top_bit(anmag) - 5);
+    anmant = (anmag == 0)  ?  32  :  (anexp >= 0)  ?  (anmag >> anexp)  :  (anmag << -anexp);
+    wanexp = anexp + ((srn >> 6) & 0xF) - 13;
+
+    wanmant = (anmant*(srn & 0x3F) + 0x30) >> 4;
+    retval = (wanexp >= 0)  ?  ((wanmant << wanexp) & 0x7FFF)  :  (wanmant >> -wanexp);
+
+    return (((an ^ srn) < 0)  ?  -retval  :  retval);
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Compute the estimated signal from the 6-zero predictor.
+ */
+static __inline__ int16_t predictor_zero(g726_state_t *s)
+{
+    int i;
+    int sezi;
+
+    sezi = fmult(s->b[0] >> 2, s->dq[0]);
+    /* ACCUM */
+    for (i = 1;  i < 6;  i++)
+        sezi += fmult(s->b[i] >> 2, s->dq[i]);
+    return (int16_t) sezi;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Computes the estimated signal from the 2-pole predictor.
+ */
+static __inline__ int16_t predictor_pole(g726_state_t *s)
+{
+    return (fmult(s->a[1] >> 2, s->sr[1]) + fmult(s->a[0] >> 2, s->sr[0]));
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Computes the quantization step size of the adaptive quantizer.
+ */
+static int step_size(g726_state_t *s)
+{
+    int y;
+    int dif;
+    int al;
+
+    if (s->ap >= 256)
+        return s->yu;
+    y = s->yl >> 6;
+    dif = s->yu - y;
+    al = s->ap >> 2;
+    if (dif > 0)
+        y += (dif*al) >> 6;
+    else if (dif < 0)
+        y += (dif*al + 0x3F) >> 6;
+    return y;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Given a raw sample, 'd', of the difference signal and a
+ * quantization step size scale factor, 'y', this routine returns the
+ * ADPCM codeword to which that sample gets quantized.  The step
+ * size scale factor division operation is done in the log base 2 domain
+ * as a subtraction.
+ */
+static int16_t quantize(int d,                  /* Raw difference signal sample */
+                        int y,                  /* Step size multiplier */
+                        const int table[],     /* quantization table */
+                        int quantizer_states)   /* table size of int16_t integers */
+{
+    int16_t dqm;    /* Magnitude of 'd' */
+    int16_t exp;    /* Integer part of base 2 log of 'd' */
+    int16_t mant;   /* Fractional part of base 2 log */
+    int16_t dl;     /* Log of magnitude of 'd' */
+    int16_t dln;    /* Step size scale factor normalized log */
+    int i;
+    int size;
+
+    /*
+     * LOG
+     *
+     * Compute base 2 log of 'd', and store in 'dl'.
+     */
+    dqm = (int16_t) abs(d);
+    exp = (int16_t) (top_bit(dqm >> 1) + 1);
+    /* Fractional portion. */
+    mant = ((dqm << 7) >> exp) & 0x7F;
+    dl = (exp << 7) + mant;
+
+    /*
+     * SUBTB
+     *
+     * "Divide" by step size multiplier.
+     */
+    dln = dl - (int16_t) (y >> 2);
+
+    /*
+     * QUAN
+     *
+     * Search for codword i for 'dln'.
+     */
+    size = (quantizer_states - 1) >> 1;
+    for (i = 0;  i < size;  i++)
+    {
+        if (dln < table[i])
+            break;
+    }
+    if (d < 0)
+    {
+        /* Take 1's complement of i */
+        return (int16_t) ((size << 1) + 1 - i);
+    }
+    if (i == 0  &&  (quantizer_states & 1))
+    {
+        /* Zero is only valid if there are an even number of states, so
+           take the 1's complement if the code is zero. */
+        return (int16_t) quantizer_states;
+    }
+    return (int16_t) i;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Returns reconstructed difference signal 'dq' obtained from
+ * codeword 'i' and quantization step size scale factor 'y'.
+ * Multiplication is performed in log base 2 domain as addition.
+ */
+static int16_t reconstruct(int sign,    /* 0 for non-negative value */
+                           int dqln,    /* G.72x codeword */
+                           int y)       /* Step size multiplier */
+{
+    int16_t dql;    /* Log of 'dq' magnitude */
+    int16_t dex;    /* Integer part of log */
+    int16_t dqt;
+    int16_t dq;     /* Reconstructed difference signal sample */
+
+    dql = (int16_t) (dqln + (y >> 2));  /* ADDA */
+
+    if (dql < 0)
+        return ((sign)  ?  -0x8000  :  0);
+    /* ANTILOG */
+    dex = (dql >> 7) & 15;
+    dqt = 128 + (dql & 127);
+    dq = (dqt << 7) >> (14 - dex);
+    return ((sign)  ?  (dq - 0x8000)  :  dq);
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * updates the state variables for each output code
+ */
+static void update(g726_state_t *s,
+                   int y,       /* quantizer step size */
+                   int wi,      /* scale factor multiplier */
+                   int fi,      /* for long/short term energies */
+                   int dq,      /* quantized prediction difference */
+                   int sr,      /* reconstructed signal */
+                   int dqsez)   /* difference from 2-pole predictor */
+{
+    int16_t mag;
+    int16_t exp;
+    int16_t a2p;        /* LIMC */
+    int16_t a1ul;       /* UPA1 */
+    int16_t pks1;       /* UPA2 */
+    int16_t fa1;
+    int16_t ylint;
+    int16_t dqthr;
+    int16_t ylfrac;
+    int16_t thr;
+    int16_t pk0;
+    int i;
+    int tr;
+
+    a2p = 0;
+    /* Needed in updating predictor poles */
+    pk0 = (dqsez < 0)  ?  1  :  0;
+
+    /* prediction difference magnitude */
+    mag = (int16_t) (dq & 0x7FFF);
+    /* TRANS */
+    ylint = (int16_t) (s->yl >> 15);            /* exponent part of yl */
+    ylfrac = (int16_t) ((s->yl >> 10) & 0x1F);  /* fractional part of yl */
+    /* Limit threshold to 31 << 10 */
+    thr = (ylint > 9)  ?  (31 << 10)  :  ((32 + ylfrac) << ylint);
+    dqthr = (thr + (thr >> 1)) >> 1;            /* dqthr = 0.75 * thr */
+    if (!s->td)                                 /* signal supposed voice */
+        tr = FALSE;
+    else if (mag <= dqthr)                      /* supposed data, but small mag */
+        tr = FALSE;                             /* treated as voice */
+    else                                        /* signal is data (modem) */
+        tr = TRUE;
+
+    /*
+     * Quantizer scale factor adaptation.
+     */
+
+    /* FUNCTW & FILTD & DELAY */
+    /* update non-steady state step size multiplier */
+    s->yu = (int16_t) (y + ((wi - y) >> 5));
+
+    /* LIMB */
+    if (s->yu < 544)
+        s->yu = 544;
+    else if (s->yu > 5120)
+        s->yu = 5120;
+
+    /* FILTE & DELAY */
+    /* update steady state step size multiplier */
+    s->yl += s->yu + ((-s->yl) >> 6);
+
+    /*
+     * Adaptive predictor coefficients.
+     */
+    if (tr)
+    {
+        /* Reset the a's and b's for a modem signal */
+        s->a[0] = 0;
+        s->a[1] = 0;
+        s->b[0] = 0;
+        s->b[1] = 0;
+        s->b[2] = 0;
+        s->b[3] = 0;
+        s->b[4] = 0;
+        s->b[5] = 0;
+    }
+    else
+    {
+        /* Update the a's and b's */
+        /* UPA2 */
+        pks1 = pk0 ^ s->pk[0];
+
+        /* Update predictor pole a[1] */
+        a2p = s->a[1] - (s->a[1] >> 7);
+        if (dqsez != 0)
+        {
+            fa1 = (pks1)  ?  s->a[0]  :  -s->a[0];
+            /* a2p = function of fa1 */
+            if (fa1 < -8191)
+                a2p -= 0x100;
+            else if (fa1 > 8191)
+                a2p += 0xFF;
+            else
+                a2p += fa1 >> 5;
+
+            if (pk0 ^ s->pk[1])
+            {
+                /* LIMC */
+                if (a2p <= -12160)
+                    a2p = -12288;
+                else if (a2p >= 12416)
+                    a2p = 12288;
+                else
+                    a2p -= 0x80;
+            }
+            else if (a2p <= -12416)
+                a2p = -12288;
+            else if (a2p >= 12160)
+                a2p = 12288;
+            else
+                a2p += 0x80;
+        }
+
+        /* TRIGB & DELAY */
+        s->a[1] = a2p;
+
+        /* UPA1 */
+        /* Update predictor pole a[0] */
+        s->a[0] -= s->a[0] >> 8;
+        if (dqsez != 0)
+        {
+            if (pks1 == 0)
+                s->a[0] += 192;
+            else
+                s->a[0] -= 192;
+        }
+        /* LIMD */
+        a1ul = 15360 - a2p;
+        if (s->a[0] < -a1ul)
+            s->a[0] = -a1ul;
+        else if (s->a[0] > a1ul)
+            s->a[0] = a1ul;
+
+        /* UPB : update predictor zeros b[6] */
+        for (i = 0;  i < 6;  i++)
+        {
+            /* Distinguish 40Kbps mode from the others */
+            s->b[i] -= s->b[i] >> ((s->bits_per_sample == 5)  ?  9  :  8);
+            if (dq & 0x7FFF)
+            {
+                /* XOR */
+                if ((dq ^ s->dq[i]) >= 0)
+                    s->b[i] += 128;
+                else
+                    s->b[i] -= 128;
+            }
+        }
+    }
+
+    for (i = 5;  i > 0;  i--)
+        s->dq[i] = s->dq[i - 1];
+    /* FLOAT A : convert dq[0] to 4-bit exp, 6-bit mantissa f.p. */
+    if (mag == 0)
+    {
+        s->dq[0] = (dq >= 0)  ?  0x20  :  0xFC20;
+    }
+    else
+    {
+        exp = (int16_t) (top_bit(mag) + 1);
+        s->dq[0] = (dq >= 0)
+                 ?  ((exp << 6) + ((mag << 6) >> exp))
+                 :  ((exp << 6) + ((mag << 6) >> exp) - 0x400);
+    }
+
+    s->sr[1] = s->sr[0];
+    /* FLOAT B : convert sr to 4-bit exp., 6-bit mantissa f.p. */
+    if (sr == 0)
+    {
+        s->sr[0] = 0x20;
+    }
+    else if (sr > 0)
+    {
+        exp = (int16_t) (top_bit(sr) + 1);
+        s->sr[0] = (int16_t) ((exp << 6) + ((sr << 6) >> exp));
+    }
+    else if (sr > -32768)
+    {
+        mag = (int16_t) -sr;
+        exp = (int16_t) (top_bit(mag) + 1);
+        s->sr[0] =  (exp << 6) + ((mag << 6) >> exp) - 0x400;
+    }
+    else
+    {
+        s->sr[0] = (uint16_t) 0xFC20;
+    }
+
+    /* DELAY A */
+    s->pk[1] = s->pk[0];
+    s->pk[0] = pk0;
+
+    /* TONE */
+    if (tr)                 /* this sample has been treated as data */
+        s->td = FALSE;      /* next one will be treated as voice */
+    else if (a2p < -11776)  /* small sample-to-sample correlation */
+        s->td = TRUE;       /* signal may be data */
+    else                    /* signal is voice */
+        s->td = FALSE;
+
+    /* Adaptation speed control. */
+    /* FILTA */
+    s->dms += ((int16_t) fi - s->dms) >> 5;
+    /* FILTB */
+    s->dml += (((int16_t) (fi << 2) - s->dml) >> 7);
+
+    if (tr)
+        s->ap = 256;
+    else if (y < 1536)                      /* SUBTC */
+        s->ap += (0x200 - s->ap) >> 4;
+    else if (s->td)
+        s->ap += (0x200 - s->ap) >> 4;
+    else if (abs((s->dms << 2) - s->dml) >= (s->dml >> 3))
+        s->ap += (0x200 - s->ap) >> 4;
+    else
+        s->ap += (-s->ap) >> 4;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int16_t tandem_adjust_alaw(int16_t sr,   /* decoder output linear PCM sample */
+                                  int se,       /* predictor estimate sample */
+                                  int y,        /* quantizer step size */
+                                  int i,        /* decoder input code */
+                                  int sign,
+                                  const int qtab[],
+                                  int quantizer_states)
+{
+    uint8_t sp; /* A-law compressed 8-bit code */
+    int16_t dx; /* prediction error */
+    int id;     /* quantized prediction error */
+    int sd;     /* adjusted A-law decoded sample value */
+
+    if (sr <= -32768)
+        sr = -1;
+    sp = linear_to_alaw((sr >> 1) << 3);
+    /* 16-bit prediction error */
+    dx = (int16_t) ((alaw_to_linear(sp) >> 2) - se);
+    id = quantize(dx, y, qtab, quantizer_states);
+    if (id == i)
+    {
+        /* No adjustment of sp required */
+        return (int16_t) sp;
+    }
+    /* sp adjustment needed */
+    /* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
+    /* 2's complement to biased unsigned */
+    if ((id ^ sign) > (i ^ sign))
+    {
+        /* sp adjusted to next lower value */
+        if (sp & 0x80)
+            sd = (sp == 0xD5)  ?  0x55  :  (((sp ^ 0x55) - 1) ^ 0x55);
+        else
+            sd = (sp == 0x2A)  ?  0x2A  :  (((sp ^ 0x55) + 1) ^ 0x55);
+    }
+    else
+    {
+        /* sp adjusted to next higher value */
+        if (sp & 0x80)
+            sd = (sp == 0xAA)  ?  0xAA  :  (((sp ^ 0x55) + 1) ^ 0x55);
+        else
+            sd = (sp == 0x55)  ?  0xD5  :  (((sp ^ 0x55) - 1) ^ 0x55);
+    }
+    return (int16_t) sd;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int16_t tandem_adjust_ulaw(int16_t sr,   /* decoder output linear PCM sample */
+                                  int se,       /* predictor estimate sample */
+                                  int y,        /* quantizer step size */
+                                  int i,        /* decoder input code */
+                                  int sign,
+                                  const int qtab[],
+                                  int quantizer_states)
+{
+    uint8_t sp; /* u-law compressed 8-bit code */
+    int16_t dx; /* prediction error */
+    int id;     /* quantized prediction error */
+    int sd;     /* adjusted u-law decoded sample value */
+
+    if (sr <= -32768)
+        sr = 0;
+    sp = linear_to_ulaw(sr << 2);
+    /* 16-bit prediction error */
+    dx = (int16_t) ((ulaw_to_linear(sp) >> 2) - se);
+    id = quantize(dx, y, qtab, quantizer_states);
+    if (id == i)
+    {
+        /* No adjustment of sp required. */
+        return (int16_t) sp;
+    }
+    /* ADPCM codes : 8, 9, ... F, 0, 1, ... , 6, 7 */
+    /* 2's complement to biased unsigned */
+    if ((id ^ sign) > (i ^ sign))
+    {
+        /* sp adjusted to next lower value */
+        if (sp & 0x80)
+            sd = (sp == 0xFF)  ?  0x7E  :  (sp + 1);
+        else
+            sd = (sp == 0x00)  ?  0x00  :  (sp - 1);
+    }
+    else
+    {
+        /* sp adjusted to next higher value */
+        if (sp & 0x80)
+            sd = (sp == 0x80)  ?  0x80  :  (sp - 1);
+        else
+            sd = (sp == 0x7F)  ?  0xFE  :  (sp + 1);
+    }
+    return (int16_t) sd;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Encodes a linear PCM, A-law or u-law input sample and returns its 3-bit code.
+ */
+static uint8_t g726_16_encoder(g726_state_t *s, int16_t amp)
+{
+    int y;
+    int16_t sei;
+    int16_t sezi;
+    int16_t se;
+    int16_t d;
+    int16_t sr;
+    int16_t dqsez;
+    int16_t dq;
+    int16_t i;
+    
+    sezi = predictor_zero(s);
+    sei = sezi + predictor_pole(s);
+    se = sei >> 1;
+    d = amp - se;
+
+    /* Quantize prediction difference */
+    y = step_size(s);
+    i = quantize(d, y, qtab_726_16, 4);
+    dq = reconstruct(i & 2, g726_16_dqlntab[i], y);
+
+    /* Reconstruct the signal */
+    sr = (dq < 0)  ?  (se - (dq & 0x3FFF))  :  (se + dq);
+
+    /* Pole prediction difference */
+    dqsez = sr + (sezi >> 1) - se;
+    
+    update(s, y, g726_16_witab[i], g726_16_fitab[i], dq, sr, dqsez);
+    return (uint8_t) i;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Decodes a 2-bit CCITT G.726_16 ADPCM code and returns
+ * the resulting 16-bit linear PCM, A-law or u-law sample value.
+ */
+static int16_t g726_16_decoder(g726_state_t *s, uint8_t code)
+{
+    int16_t sezi;
+    int16_t sei;
+    int16_t se;
+    int16_t sr;
+    int16_t dq;
+    int16_t dqsez;
+    int y;
+
+    /* Mask to get proper bits */
+    code &= 0x03;
+    sezi = predictor_zero(s);
+    sei = sezi + predictor_pole(s);
+
+    y = step_size(s);
+    dq = reconstruct(code & 2, g726_16_dqlntab[code], y);
+
+    /* Reconstruct the signal */
+    se = sei >> 1;
+    sr = (dq < 0)  ?  (se - (dq & 0x3FFF))  :  (se + dq);
+
+    /* Pole prediction difference */
+    dqsez = sr + (sezi >> 1) - se;
+
+    update(s, y, g726_16_witab[code], g726_16_fitab[code], dq, sr, dqsez);
+
+    switch (s->ext_coding)
+    {
+    case G726_ENCODING_ALAW:
+        return tandem_adjust_alaw(sr, se, y, code, 2, qtab_726_16, 4);
+    case G726_ENCODING_ULAW:
+        return tandem_adjust_ulaw(sr, se, y, code, 2, qtab_726_16, 4);
+    }
+    return (sr << 2);
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Encodes a linear PCM, A-law or u-law input sample and returns its 3-bit code.
+ */
+static uint8_t g726_24_encoder(g726_state_t *s, int16_t amp)
+{
+    int16_t sei;
+    int16_t sezi;
+    int16_t se;
+    int16_t d;
+    int16_t sr;
+    int16_t dqsez;
+    int16_t dq;
+    int16_t i;
+    int y;
+    
+    sezi = predictor_zero(s);
+    sei = sezi + predictor_pole(s);
+    se = sei >> 1;
+    d = amp - se;
+
+    /* Quantize prediction difference */
+    y = step_size(s);
+    i = quantize(d, y, qtab_726_24, 7);
+    dq = reconstruct(i & 4, g726_24_dqlntab[i], y);
+
+    /* Reconstruct the signal */
+    sr = (dq < 0)  ?  (se - (dq & 0x3FFF))  :  (se + dq);
+
+    /* Pole prediction difference */
+    dqsez = sr + (sezi >> 1) - se;
+    
+    update(s, y, g726_24_witab[i], g726_24_fitab[i], dq, sr, dqsez);
+    return (uint8_t) i;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Decodes a 3-bit CCITT G.726_24 ADPCM code and returns
+ * the resulting 16-bit linear PCM, A-law or u-law sample value.
+ */
+static int16_t g726_24_decoder(g726_state_t *s, uint8_t code)
+{
+    int16_t sezi;
+    int16_t sei;
+    int16_t se;
+    int16_t sr;
+    int16_t dq;
+    int16_t dqsez;
+    int y;
+
+    /* Mask to get proper bits */
+    code &= 0x07;
+    sezi = predictor_zero(s);
+    sei = sezi + predictor_pole(s);
+
+    y = step_size(s);
+    dq = reconstruct(code & 4, g726_24_dqlntab[code], y);
+
+    /* Reconstruct the signal */
+    se = sei >> 1;
+    sr = (dq < 0)  ?  (se - (dq & 0x3FFF))  :  (se + dq);
+
+    /* Pole prediction difference */
+    dqsez = sr + (sezi >> 1) - se;
+
+    update(s, y, g726_24_witab[code], g726_24_fitab[code], dq, sr, dqsez);
+
+    switch (s->ext_coding)
+    {
+    case G726_ENCODING_ALAW:
+        return tandem_adjust_alaw(sr, se, y, code, 4, qtab_726_24, 7);
+    case G726_ENCODING_ULAW:
+        return tandem_adjust_ulaw(sr, se, y, code, 4, qtab_726_24, 7);
+    }
+    return (sr << 2);
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Encodes a linear input sample and returns its 4-bit code.
+ */
+static uint8_t g726_32_encoder(g726_state_t *s, int16_t amp)
+{
+    int16_t sei;
+    int16_t sezi;
+    int16_t se;
+    int16_t d;
+    int16_t sr;
+    int16_t dqsez;
+    int16_t dq;
+    int16_t i;
+    int y;
+    
+    sezi = predictor_zero(s);
+    sei = sezi + predictor_pole(s);
+    se = sei >> 1;
+    d = amp - se;
+
+    /* Quantize the prediction difference */
+    y = step_size(s);
+    i = quantize(d, y, qtab_726_32, 15);
+    dq = reconstruct(i & 8, g726_32_dqlntab[i], y);
+
+    /* Reconstruct the signal */
+    sr = (dq < 0)  ?  (se - (dq & 0x3FFF))  :  (se + dq);
+
+    /* Pole prediction difference */
+    dqsez = sr + (sezi >> 1) - se;
+
+    update(s, y, g726_32_witab[i], g726_32_fitab[i], dq, sr, dqsez);
+    return (uint8_t) i;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Decodes a 4-bit CCITT G.726_32 ADPCM code and returns
+ * the resulting 16-bit linear PCM, A-law or u-law sample value.
+ */
+static int16_t g726_32_decoder(g726_state_t *s, uint8_t code)
+{
+    int16_t sezi;
+    int16_t sei;
+    int16_t se;
+    int16_t sr;
+    int16_t dq;
+    int16_t dqsez;
+    int y;
+
+    /* Mask to get proper bits */
+    code &= 0x0F;
+    sezi = predictor_zero(s);
+    sei = sezi + predictor_pole(s);
+
+    y = step_size(s);
+    dq = reconstruct(code & 8, g726_32_dqlntab[code], y);
+
+    /* Reconstruct the signal */
+    se = sei >> 1;
+    sr = (dq < 0)  ?  (se - (dq & 0x3FFF))  :  (se + dq);
+
+    /* Pole prediction difference */
+    dqsez = sr + (sezi >> 1) - se;
+
+    update(s, y, g726_32_witab[code], g726_32_fitab[code], dq, sr, dqsez);
+
+    switch (s->ext_coding)
+    {
+    case G726_ENCODING_ALAW:
+        return tandem_adjust_alaw(sr, se, y, code, 8, qtab_726_32, 15);
+    case G726_ENCODING_ULAW:
+        return tandem_adjust_ulaw(sr, se, y, code, 8, qtab_726_32, 15);
+    }
+    return (sr << 2);
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Encodes a 16-bit linear PCM, A-law or u-law input sample and retuens
+ * the resulting 5-bit CCITT G.726 40Kbps code.
+ */
+static uint8_t g726_40_encoder(g726_state_t *s, int16_t amp)
+{
+    int16_t sei;
+    int16_t sezi;
+    int16_t se;
+    int16_t d;
+    int16_t sr;
+    int16_t dqsez;
+    int16_t dq;
+    int16_t i;
+    int y;
+    
+    sezi = predictor_zero(s);
+    sei = sezi + predictor_pole(s);
+    se = sei >> 1;
+    d = amp - se;
+
+    /* Quantize prediction difference */
+    y = step_size(s);
+    i = quantize(d, y, qtab_726_40, 31);
+    dq = reconstruct(i & 0x10, g726_40_dqlntab[i], y);
+
+    /* Reconstruct the signal */
+    sr = (dq < 0)  ?  (se - (dq & 0x7FFF))  :  (se + dq);
+
+    /* Pole prediction difference */
+    dqsez = sr + (sezi >> 1) - se;
+
+    update(s, y, g726_40_witab[i], g726_40_fitab[i], dq, sr, dqsez);
+    return (uint8_t) i;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * Decodes a 5-bit CCITT G.726 40Kbps code and returns
+ * the resulting 16-bit linear PCM, A-law or u-law sample value.
+ */
+static int16_t g726_40_decoder(g726_state_t *s, uint8_t code)
+{
+    int16_t sezi;
+    int16_t sei;
+    int16_t se;
+    int16_t sr;
+    int16_t dq;
+    int16_t dqsez;
+    int y;
+
+    /* Mask to get proper bits */
+    code &= 0x1F;
+    sezi = predictor_zero(s);
+    sei = sezi + predictor_pole(s);
+        
+    y = step_size(s);
+    dq = reconstruct(code & 0x10, g726_40_dqlntab[code], y);
+
+    /* Reconstruct the signal */
+    se = sei >> 1;
+    sr = (dq < 0)  ?  (se - (dq & 0x7FFF))  :  (se + dq);
+
+    /* Pole prediction difference */
+    dqsez = sr + (sezi >> 1) - se;
+
+    update(s, y, g726_40_witab[code], g726_40_fitab[code], dq, sr, dqsez);
+
+    switch (s->ext_coding)
+    {
+    case G726_ENCODING_ALAW:
+        return tandem_adjust_alaw(sr, se, y, code, 0x10, qtab_726_40, 31);
+    case G726_ENCODING_ULAW:
+        return tandem_adjust_ulaw(sr, se, y, code, 0x10, qtab_726_40, 31);
+    }
+    return (sr << 2);
+}
+/*- End of function --------------------------------------------------------*/
+
+g726_state_t *g726_init(g726_state_t *s, int bit_rate, int ext_coding, int packing)
+{
+    int i;
+
+    if (bit_rate != 16000  &&  bit_rate != 24000  &&  bit_rate != 32000  &&  bit_rate != 40000)
+        return NULL;
+    if (s == NULL)
+    {
+        if ((s = (g726_state_t *) malloc(sizeof(*s))) == NULL)
+            return  NULL;
+    }
+    s->yl = 34816;
+    s->yu = 544;
+    s->dms = 0;
+    s->dml = 0;
+    s->ap = 0;
+    s->rate = bit_rate;
+    s->ext_coding = ext_coding;
+    s->packing = packing;
+    for (i = 0; i < 2; i++)
+    {
+        s->a[i] = 0;
+        s->pk[i] = 0;
+        s->sr[i] = 32;
+    }
+    for (i = 0; i < 6; i++)
+    {
+        s->b[i] = 0;
+        s->dq[i] = 32;
+    }
+    s->td = FALSE;
+    switch (bit_rate)
+    {
+    case 16000:
+        s->enc_func = g726_16_encoder;
+        s->dec_func = g726_16_decoder;
+        s->bits_per_sample = 2;
+        break;
+    case 24000:
+        s->enc_func = g726_24_encoder;
+        s->dec_func = g726_24_decoder;
+        s->bits_per_sample = 3;
+        break;
+    case 32000:
+    default:
+        s->enc_func = g726_32_encoder;
+        s->dec_func = g726_32_decoder;
+        s->bits_per_sample = 4;
+        break;
+    case 40000:
+        s->enc_func = g726_40_encoder;
+        s->dec_func = g726_40_decoder;
+        s->bits_per_sample = 5;
+        break;
+    }
+    bitstream_init(&s->bs);
+    return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int g726_release(g726_state_t *s)
+{
+    free(s);
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+int g726_decode(g726_state_t *s,
+                int16_t amp[],
+                const uint8_t g726_data[],
+                int g726_bytes)
+{
+    int i;
+    int samples;
+    uint8_t code;
+    int sl;
+
+    for (samples = i = 0;  ;  )
+    {
+        if (s->packing != G726_PACKING_NONE)
+        {
+            /* Unpack the code bits */
+            if (s->packing != G726_PACKING_LEFT)
+            {
+                if (s->bs.residue < s->bits_per_sample)
+                {
+                    if (i >= g726_bytes)
+                        break;
+                    s->bs.bitstream |= (g726_data[i++] << s->bs.residue);
+                    s->bs.residue += 8;
+                }
+                code = (uint8_t) (s->bs.bitstream & ((1 << s->bits_per_sample) - 1));
+                s->bs.bitstream >>= s->bits_per_sample;
+            }
+            else
+            {
+                if (s->bs.residue < s->bits_per_sample)
+                {
+                    if (i >= g726_bytes)
+                        break;
+                    s->bs.bitstream = (s->bs.bitstream << 8) | g726_data[i++];
+                    s->bs.residue += 8;
+                }
+                code = (uint8_t) ((s->bs.bitstream >> (s->bs.residue - s->bits_per_sample)) & ((1 << s->bits_per_sample) - 1));
+            }
+            s->bs.residue -= s->bits_per_sample;
+        }
+        else
+        {
+            if (i >= g726_bytes)
+                break;
+            code = g726_data[i++];
+        }
+        sl = s->dec_func(s, code);
+        if (s->ext_coding != G726_ENCODING_LINEAR)
+            ((uint8_t *) amp)[samples++] = (uint8_t) sl;
+        else
+            amp[samples++] = (int16_t) sl;
+    }
+    return samples;
+}
+/*- End of function --------------------------------------------------------*/
+
+int g726_encode(g726_state_t *s,
+                uint8_t g726_data[],
+                const int16_t amp[],
+                int len)
+{
+    int i;
+    int g726_bytes;
+    int16_t sl;
+    uint8_t code;
+
+    for (g726_bytes = i = 0;  i < len;  i++)
+    {
+        /* Linearize the input sample to 14-bit PCM */
+        switch (s->ext_coding)
+        {
+        case G726_ENCODING_ALAW:
+            sl = alaw_to_linear(((const uint8_t *) amp)[i]) >> 2;
+            break;
+        case G726_ENCODING_ULAW:
+            sl = ulaw_to_linear(((const uint8_t *) amp)[i]) >> 2;
+            break;
+        default:
+            sl = amp[i] >> 2;
+            break;
+        }
+        code = s->enc_func(s, sl);
+        if (s->packing != G726_PACKING_NONE)
+        {
+            /* Pack the code bits */
+            if (s->packing != G726_PACKING_LEFT)
+            {
+                s->bs.bitstream |= (code << s->bs.residue);
+                s->bs.residue += s->bits_per_sample;
+                if (s->bs.residue >= 8)
+                {
+                    g726_data[g726_bytes++] = (uint8_t) (s->bs.bitstream & 0xFF);
+                    s->bs.bitstream >>= 8;
+                    s->bs.residue -= 8;
+                }
+            }
+            else
+            {
+                s->bs.bitstream = (s->bs.bitstream << s->bits_per_sample) | code;
+                s->bs.residue += s->bits_per_sample;
+                if (s->bs.residue >= 8)
+                {
+                    g726_data[g726_bytes++] = (uint8_t) ((s->bs.bitstream >> (s->bs.residue - 8)) & 0xFF);
+                    s->bs.residue -= 8;
+                }
+            }
+        }
+        else
+        {
+            g726_data[g726_bytes++] = (uint8_t) code;
+        }
+    }
+    return g726_bytes;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/gsm0610_decode.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/gsm0610_decode.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,358 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * gsm0610_decode.c - GSM 06.10 full rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the widely used GSM 06.10 code available from
+ * http://kbs.cs.tu-berlin.de/~jutta/toast.html
+ *
+ * $Id: gsm0610_decode.c,v 1.14 2007/08/21 14:25:54 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <assert.h>
+#include <inttypes.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+#include <stdlib.h>
+#include <memory.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/bitstream.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/gsm0610.h"
+
+#include "gsm0610_local.h"
+
+/* 4.3 FIXED POINT IMPLEMENTATION OF THE RPE-LTP DECODER */
+
+static void postprocessing(gsm0610_state_t *s, int16_t amp[])
+{
+    int k;
+    int16_t msr;
+    int16_t tmp;
+
+    msr = s->msr;
+    for (k = 0;  k < GSM0610_FRAME_LEN;  k++)
+    {
+        tmp = gsm_mult_r(msr, 28180);
+        /* De-emphasis */
+        msr = gsm_add(amp[k], tmp);
+        /* Truncation & upscaling */
+        amp[k] = (int16_t) (gsm_add(msr, msr) & 0xFFF8);
+    }
+    /*endfor*/
+    s->msr = msr;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void decode_a_frame(gsm0610_state_t *s,
+                           int16_t amp[GSM0610_FRAME_LEN],
+                           gsm0610_frame_t *f)
+{
+    int j;
+    int k;
+    int16_t erp[40];
+    int16_t wt[GSM0610_FRAME_LEN];
+    int16_t *drp;
+
+    drp = s->dp0 + 120;
+    for (j = 0;  j < 4;  j++)
+    {
+        gsm0610_rpe_decoding(s, f->xmaxc[j], f->Mc[j], f->xMc[j], erp);
+        gsm0610_long_term_synthesis_filtering(s, f->Nc[j], f->bc[j], erp, drp);
+        for (k = 0;  k < 40;  k++)
+            wt[j*40 + k] = drp[k];
+        /*endfor*/
+    }
+    /*endfor*/
+
+    gsm0610_short_term_synthesis_filter(s, f->LARc, wt, amp);
+    postprocessing(s, amp);
+}
+/*- End of function --------------------------------------------------------*/
+
+int gsm0610_unpack_none(gsm0610_frame_t *s, const uint8_t c[])
+{
+    int i;
+    int j;
+    int k;
+    
+    i = 0;
+    for (j = 0;  j < 8;  j++)
+        s->LARc[j] = c[i++];
+    for (j = 0;  j < 4;  j++)
+    {
+        s->Nc[j] = c[i++];
+        s->bc[j] = c[i++];
+        s->Mc[j] = c[i++];
+        s->xmaxc[j] = c[i++];
+        for (k = 0;  k < 13;  k++)
+            s->xMc[j][k] = c[i++];
+    }
+    return 76;
+}
+/*- End of function --------------------------------------------------------*/
+
+int gsm0610_unpack_wav49(gsm0610_frame_t *s, const uint8_t c[])
+{
+    uint16_t sr;
+    int i;
+
+    sr = *c++;
+    s->LARc[0] = sr & 0x3F;
+    sr >>= 6;
+    sr |= (uint16_t) *c++ << 2;
+    s->LARc[1] = sr & 0x3F;
+    sr >>= 6;
+    sr |= (uint16_t) *c++ << 4;
+    s->LARc[2] = sr & 0x1F;
+    sr >>= 5;
+    s->LARc[3] = sr & 0x1F;
+    sr >>= 5;
+    sr |= (uint16_t) *c++ << 2;
+    s->LARc[4] = sr & 0xF;
+    sr >>= 4;
+    s->LARc[5] = sr & 0xF;
+    sr >>= 4;
+    sr |= (uint16_t) *c++ << 2;
+    s->LARc[6] = sr & 0x7;
+    sr >>= 3;
+    s->LARc[7] = sr & 0x7;
+    sr >>= 3;
+
+    for (i = 0;  i < 4;  i++)
+    {
+        sr |= (uint16_t) *c++ << 4;
+        s->Nc[i] = sr & 0x7F;
+        sr >>= 7;
+        s->bc[i] = sr & 0x3;
+        sr >>= 2;
+        s->Mc[i] = sr & 0x3;
+        sr >>= 2;
+        sr |= (uint16_t) *c++ << 1;
+        s->xmaxc[i] = sr & 0x3F;
+        sr >>= 6;
+        s->xMc[i][0] = sr & 0x7;
+        sr >>= 3;
+        sr = *c++;
+        s->xMc[i][1] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][2] = sr & 0x7;
+        sr >>= 3;
+        sr |= (uint16_t) *c++ << 2;
+        s->xMc[i][3] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][4] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][5] = sr & 0x7;
+        sr >>= 3;
+        sr |= (uint16_t) *c++ << 1;
+        s->xMc[i][6] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][7] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][8] = sr & 0x7;
+        sr >>= 3;
+        sr = *c++;
+        s->xMc[i][9] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][10] = sr & 0x7;
+        sr >>= 3;
+        sr |= (uint16_t) *c++ << 2;
+        s->xMc[i][11] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][12] = sr & 0x7;
+        sr >>= 3;
+    }
+
+    s++;
+    sr |= (uint16_t) *c++ << 4;
+    s->LARc[0] = sr & 0x3F;
+    sr >>= 6;
+    s->LARc[1] = sr & 0x3F;
+    sr >>= 6;
+    sr = *c++;
+    s->LARc[2] = sr & 0x1F;
+    sr >>= 5;
+    sr |= (uint16_t) *c++ << 3;
+    s->LARc[3] = sr & 0x1F;
+    sr >>= 5;
+    s->LARc[4] = sr & 0xF;
+    sr >>= 4;
+    sr |= (uint16_t) *c++ << 2;
+    s->LARc[5] = sr & 0xF;
+    sr >>= 4;
+    s->LARc[6] = sr & 0x7;
+    sr >>= 3;
+    s->LARc[7] = sr & 0x7;
+    sr >>= 3;
+
+    for (i = 0;  i < 4;  i++)
+    {
+        sr = *c++;
+        s->Nc[i] = sr & 0x7F;
+        sr >>= 7;
+        sr |= (uint16_t) *c++ << 1;
+        s->bc[i] = sr & 0x3;
+        sr >>= 2;
+        s->Mc[i] = sr & 0x3;
+        sr >>= 2;
+        sr |= (uint16_t) *c++ << 5;
+        s->xmaxc[i] = sr & 0x3F;
+        sr >>= 6;
+        s->xMc[i][0] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][1] = sr & 0x7;
+        sr >>= 3;
+        sr |= (uint16_t) *c++ << 1;
+        s->xMc[i][2] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][3] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][4] = sr & 0x7;
+        sr >>= 3;
+        sr = *c++;
+        s->xMc[i][5] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][6] = sr & 0x7;
+        sr >>= 3;
+        sr |= (uint16_t) *c++ << 2;
+        s->xMc[i][7] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][8] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][9] = sr & 0x7;
+        sr >>= 3;
+        sr |= (uint16_t) *c++ << 1;
+        s->xMc[i][10] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][11] = sr & 0x7;
+        sr >>= 3;
+        s->xMc[i][12] = sr & 0x7;
+        sr >>= 3;
+    }
+    return 65;
+}
+/*- End of function --------------------------------------------------------*/
+
+int gsm0610_unpack_voip(gsm0610_frame_t *s, const uint8_t c[33])
+{
+    int i;
+
+    s->LARc[0]  = (*c++ & 0xF) << 2;
+    s->LARc[0] |= (*c >> 6) & 0x3;
+    s->LARc[1]  = *c++ & 0x3F;
+    s->LARc[2]  = (*c >> 3) & 0x1F;
+    s->LARc[3]  = (*c++ & 0x7) << 2;
+    s->LARc[3] |= (*c >> 6) & 0x3;
+    s->LARc[4]  = (*c >> 2) & 0xF;
+    s->LARc[5]  = (*c++ & 0x3) << 2;
+    s->LARc[5] |= (*c >> 6) & 0x3;
+    s->LARc[6]  = (*c >> 3) & 0x7;
+    s->LARc[7]  = *c++ & 0x7;
+
+    for (i = 0;  i < 4;  i++)
+    {
+        s->Nc[i]       = (*c >> 1) & 0x7F;
+        s->bc[i]       = (*c++ & 0x1) << 1;
+        s->bc[i]      |= (*c >> 7) & 0x1;
+        s->Mc[i]       = (*c >> 5) & 0x3;
+        s->xmaxc[i]    = (*c++ & 0x1F) << 1;
+        s->xmaxc[i]   |= (*c >> 7) & 0x1;
+        s->xMc[i][0]   = (*c >> 4) & 0x7;
+        s->xMc[i][1]   = (*c >> 1) & 0x7;
+        s->xMc[i][2]   = (*c++ & 0x1) << 2;
+        s->xMc[i][2]  |= (*c >> 6) & 0x3;
+        s->xMc[i][3]   = (*c >> 3) & 0x7;
+        s->xMc[i][4]   = *c++ & 0x7;
+        s->xMc[i][5]   = (*c >> 5) & 0x7;
+        s->xMc[i][6]   = (*c >> 2) & 0x7;
+        s->xMc[i][7]   = (*c++ & 0x3) << 1;
+        s->xMc[i][7]  |= (*c >> 7) & 0x1;
+        s->xMc[i][8]   = (*c >> 4) & 0x7;
+        s->xMc[i][9]   = (*c >> 1) & 0x7;
+        s->xMc[i][10]  = (*c++ & 0x1) << 2;
+        s->xMc[i][10] |= (*c >> 6) & 0x3;
+        s->xMc[i][11]  = (*c >> 3) & 0x7;
+        s->xMc[i][12]  = *c++ & 0x7;
+    }
+    return 33;
+}
+/*- End of function --------------------------------------------------------*/
+
+int gsm0610_decode(gsm0610_state_t *s, int16_t amp[], const uint8_t code[], int quant)
+{
+    gsm0610_frame_t frame[2];
+    const uint8_t *c;
+    int bytes;
+    int i;
+
+    c = code;
+    for (i = 0;  i < quant;  i++)
+    {
+        switch (s->packing)
+        {
+        default:
+        case GSM0610_PACKING_NONE:
+            if ((bytes = gsm0610_unpack_none(frame, c)) >= 0)
+            {
+                decode_a_frame(s, amp, frame);
+                amp += GSM0610_FRAME_LEN;
+            }
+            break;
+        case GSM0610_PACKING_WAV49:
+            if ((bytes = gsm0610_unpack_wav49(frame, c)) >= 0)
+            {
+                decode_a_frame(s, amp, frame);
+                amp += GSM0610_FRAME_LEN;
+                decode_a_frame(s, amp, frame + 1);
+                amp += GSM0610_FRAME_LEN;
+            }
+            break;
+        case GSM0610_PACKING_VOIP:
+            if ((bytes = gsm0610_unpack_voip(frame, c)) >= 0)
+            {
+                decode_a_frame(s, amp, frame);
+                amp += GSM0610_FRAME_LEN;
+            }
+            break;
+        }
+        /*endswitch*/
+        if (bytes < 0)
+            return 0;
+        c += bytes;
+    }
+    /*endwhile*/
+    return quant*GSM0610_FRAME_LEN;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/gsm0610_encode.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/gsm0610_encode.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,334 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * gsm0610_encode.c - GSM 06.10 full rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the widely used GSM 06.10 code available from
+ * http://kbs.cs.tu-berlin.de/~jutta/toast.html
+ *
+ * $Id: gsm0610_encode.c,v 1.17 2007/11/26 13:28:59 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <assert.h>
+#include <inttypes.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+#include <stdlib.h>
+#include <memory.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/bitstream.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/gsm0610.h"
+
+#include "gsm0610_local.h"
+
+/* 4.2 FIXED POINT IMPLEMENTATION OF THE RPE-LTP CODER */
+
+/* The RPE-LTD coder works on a frame by frame basis.  The length of
+   the frame is equal to 160 samples.  Some computations are done
+   once per frame to produce at the output of the coder the
+   LARc[1..8] parameters which are the coded LAR coefficients and 
+   also to realize the inverse filtering operation for the entire
+   frame (160 samples of signal d[0..159]).  These parts produce at
+   the output of the coder:
+
+   Procedure 4.2.11 to 4.2.18 are to be executed four times per
+   frame.  That means once for each sub-segment RPE-LTP analysis of
+   40 samples.  These parts produce at the output of the coder.
+*/
+static void encode_a_frame(gsm0610_state_t *s, gsm0610_frame_t *f, const int16_t amp[])
+{
+    int k;
+    int16_t *dp;
+    int16_t *dpp;
+    int16_t so[GSM0610_FRAME_LEN];
+    int i;
+
+    dp = s->dp0 + 120;
+    dpp = dp;
+    gsm0610_preprocess(s, amp, so);
+    gsm0610_lpc_analysis(s, so, f->LARc);
+    gsm0610_short_term_analysis_filter(s, f->LARc, so);
+
+    for (k = 0;  k < 4;  k++)
+    {
+        gsm0610_long_term_predictor(s,
+                                    so + k*40,
+                                    dp,
+                                    s->e + 5,
+                                    dpp,
+                                    &f->Nc[k],
+                                    &f->bc[k]);
+        gsm0610_rpe_encoding(s, s->e + 5, &f->xmaxc[k], &f->Mc[k], f->xMc[k]);
+
+        for (i = 0;  i < 40;  i++)
+            dp[i] = gsm_add(s->e[5 + i], dpp[i]);
+        /*endfor*/
+        dp += 40;
+        dpp += 40;
+    }
+    /*endfor*/
+    memcpy((char *) s->dp0,
+           (char *) (s->dp0 + GSM0610_FRAME_LEN),
+           120*sizeof(*s->dp0));
+}
+/*- End of function --------------------------------------------------------*/
+
+gsm0610_state_t *gsm0610_init(gsm0610_state_t *s, int packing)
+{
+    if (s == NULL)
+    {
+        s = (gsm0610_state_t *) malloc(sizeof (*s));
+        if (s == NULL)
+            return  NULL;
+        /*endif*/
+    }
+    /*endif*/
+    memset((char *) s, '\0', sizeof (gsm0610_state_t));
+    s->nrp = 40;
+    s->packing = packing;
+    return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int gsm0610_set_packing(gsm0610_state_t *s, int packing)
+{
+    s->packing = packing;
+    return 0;    
+}
+/*- End of function --------------------------------------------------------*/
+
+int gsm0610_release(gsm0610_state_t *s)
+{
+    if (s)
+        free(s);
+    /*endif*/
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+int gsm0610_pack_none(uint8_t c[], const gsm0610_frame_t *s)
+{
+    int i;
+    int j;
+    int k;
+    
+    i = 0;
+    for (j = 0;  j < 8;  j++)
+        c[i++] = (uint8_t) s->LARc[j];
+    for (j = 0;  j < 4;  j++)
+    {
+        c[i++] = (uint8_t) s->Nc[j];
+        c[i++] = (uint8_t) s->bc[j];
+        c[i++] = (uint8_t) s->Mc[j];
+        c[i++] = (uint8_t) s->xmaxc[j];
+        for (k = 0;  k < 13;  k++)
+            c[i++] = (uint8_t) s->xMc[j][k];
+    }
+    return 76;
+}
+/*- End of function --------------------------------------------------------*/
+
+int gsm0610_pack_wav49(uint8_t c[], const gsm0610_frame_t *s)
+{
+    uint16_t sr;
+    int i;
+ 
+	sr = 0;
+	sr = (sr >> 6) | (s->LARc[0] << 10);
+	sr = (sr >> 6) | (s->LARc[1] << 10);
+	*c++ = sr >> 4;
+	sr = (sr >> 5) | (s->LARc[2] << 11);
+	*c++ = sr >> 7;
+	sr = (sr >> 5) | (s->LARc[3] << 11);
+	sr = (sr >> 4) | (s->LARc[4] << 12);
+	*c++ = sr >> 6;
+	sr = (sr >> 4) | (s->LARc[5] << 12);
+	sr = (sr >> 3) | (s->LARc[6] << 13);
+	*c++ = sr >> 7;
+	sr = (sr >> 3) | (s->LARc[7] << 13);
+
+    for (i = 0;  i < 4;  i++)
+    {
+    	sr = (sr >> 7) | (s->Nc[i] << 9);
+    	*c++ = sr >> 5;
+    	sr = (sr >> 2) | (s->bc[i] << 14);
+    	sr = (sr >> 2) | (s->Mc[i] << 14);
+    	sr = (sr >> 6) | (s->xmaxc[i] << 10);
+    	*c++ = sr >> 3;
+    	sr = (sr >> 3) | (s->xMc[i][0] << 13);
+    	*c++ = sr >> 8;
+    	sr = (sr >> 3) | (s->xMc[i][1] << 13);
+    	sr = (sr >> 3) | (s->xMc[i][2] << 13);
+        sr = (sr >> 3) | (s->xMc[i][3] << 13);
+        *c++ = sr >> 7;
+        sr = (sr >> 3) | (s->xMc[i][4] << 13);
+        sr = (sr >> 3) | (s->xMc[i][5] << 13);
+        sr = (sr >> 3) | (s->xMc[i][6] << 13);
+        *c++ = sr >> 6;
+        sr = (sr >> 3) | (s->xMc[i][7] << 13);
+        sr = (sr >> 3) | (s->xMc[i][8] << 13);
+        *c++ = sr >> 8;
+        sr = (sr >> 3) | (s->xMc[i][9] << 13);
+        sr = (sr >> 3) | (s->xMc[i][10] << 13);
+        sr = (sr >> 3) | (s->xMc[i][11] << 13);
+        *c++ = sr >> 7;
+        sr = (sr >> 3) | (s->xMc[i][12] << 13);
+    }
+
+    s++;
+    sr = (sr >> 6) | (s->LARc[0] << 10);
+    *c++ = sr >> 6;
+    sr = (sr >> 6) | (s->LARc[1] << 10);
+    *c++ = sr >> 8;
+    sr = (sr >> 5) | (s->LARc[2] << 11);
+    sr = (sr >> 5) | (s->LARc[3] << 11);
+    *c++ = sr >> 6;
+    sr = (sr >> 4) | (s->LARc[4] << 12);
+    sr = (sr >> 4) | (s->LARc[5] << 12);
+    *c++ = sr >> 6;
+    sr = (sr >> 3) | (s->LARc[6] << 13);
+    sr = (sr >> 3) | (s->LARc[7] << 13);
+    *c++ = sr >> 8;
+
+    for (i = 0;  i < 4;  i++)
+    {
+        sr = (sr >> 7) | (s->Nc[i] << 9);
+        sr = (sr >> 2) | (s->bc[i] << 14);
+        *c++ = sr >> 7;
+        sr = (sr >> 2) | (s->Mc[i] << 14);
+        sr = (sr >> 6) | (s->xmaxc[i] << 10);
+        *c++ = sr >> 7;
+        sr = (sr >> 3) | (s->xMc[i][0] << 13);
+        sr = (sr >> 3) | (s->xMc[i][1] << 13);
+        sr = (sr >> 3) | (s->xMc[i][2] << 13);
+        *c++ = sr >> 6;
+        sr = (sr >> 3) | (s->xMc[i][3] << 13);
+        sr = (sr >> 3) | (s->xMc[i][4] << 13);
+        *c++ = sr >> 8;
+        sr = (sr >> 3) | (s->xMc[i][5] << 13);
+        sr = (sr >> 3) | (s->xMc[i][6] << 13);
+        sr = (sr >> 3) | (s->xMc[i][7] << 13);
+        *c++ = sr >> 7;
+        sr = (sr >> 3) | (s->xMc[i][8] << 13);
+        sr = (sr >> 3) | (s->xMc[i][9] << 13);
+        sr = (sr >> 3) | (s->xMc[i][10] << 13);
+        *c++ = sr >> 6;
+        sr = (sr >> 3) | (s->xMc[i][11] << 13);
+        sr = (sr >> 3) | (s->xMc[i][12] << 13);
+        *c++ = sr >> 8;
+    }
+    return 65;
+}
+/*- End of function --------------------------------------------------------*/
+
+int gsm0610_pack_voip(uint8_t c[33], const gsm0610_frame_t *s)
+{
+    int i;
+
+    *c++ = ((GSM0610_MAGIC & 0xF) << 4)
+         | ((s->LARc[0] >> 2) & 0xF);
+    *c++ = ((s->LARc[0] & 0x3) << 6)
+         |  (s->LARc[1] & 0x3F);
+    *c++ = ((s->LARc[2] & 0x1F) << 3)
+         | ((s->LARc[3] >> 2) & 0x7);
+    *c++ = ((s->LARc[3] & 0x3) << 6)
+         | ((s->LARc[4] & 0xF) << 2)
+         | ((s->LARc[5] >> 2) & 0x3);
+    *c++ = ((s->LARc[5] & 0x3) << 6)
+         | ((s->LARc[6] & 0x7) << 3)
+         |  (s->LARc[7] & 0x7);
+
+    for (i = 0;  i < 4;  i++)
+    {
+        *c++ = ((s->Nc[i] & 0x7F) << 1)
+             | ((s->bc[i] >> 1) & 0x1);
+        *c++ = ((s->bc[i] & 0x1) << 7)
+             | ((s->Mc[i] & 0x3) << 5)
+             | ((s->xmaxc[i] >> 1) & 0x1F);
+        *c++ = ((s->xmaxc[i] & 0x1) << 7)
+             | ((s->xMc[i][0] & 0x7) << 4)
+             | ((s->xMc[i][1] & 0x7) << 1)
+             | ((s->xMc[i][2] >> 2) & 0x1);
+        *c++ = ((s->xMc[i][2] & 0x3) << 6)
+             | ((s->xMc[i][3] & 0x7) << 3)
+             |  (s->xMc[i][4] & 0x7);
+        *c++ = ((s->xMc[i][5] & 0x7) << 5)
+             | ((s->xMc[i][6] & 0x7) << 2)
+             | ((s->xMc[i][7] >> 1) & 0x3);
+        *c++ = ((s->xMc[i][7] & 0x1) << 7)
+             | ((s->xMc[i][8] & 0x7) << 4)
+             | ((s->xMc[i][9] & 0x7) << 1)
+             | ((s->xMc[i][10] >> 2) & 0x1);
+        *c++ = ((s->xMc[i][10] & 0x3) << 6)
+             | ((s->xMc[i][11] & 0x7) << 3)
+             |  (s->xMc[i][12] & 0x7);
+    }
+    return 33;
+}
+/*- End of function --------------------------------------------------------*/
+
+int gsm0610_encode(gsm0610_state_t *s, uint8_t code[], const int16_t amp[], int quant)
+{
+    gsm0610_frame_t frame[2];
+    uint8_t *c;
+    int i;
+
+    c = code;
+    for (i = 0;  i < quant;  i++)
+    {
+        encode_a_frame(s, frame, amp);
+        switch (s->packing)
+        {
+        case GSM0610_PACKING_NONE:
+            c += gsm0610_pack_none(c, frame);
+            amp += GSM0610_FRAME_LEN;
+            break;
+        case GSM0610_PACKING_WAV49:
+            amp += GSM0610_FRAME_LEN;
+            encode_a_frame(s, frame + 1, amp);
+            amp += GSM0610_FRAME_LEN;
+            c += gsm0610_pack_wav49(c, frame);
+            break;
+        case GSM0610_PACKING_VOIP:
+            c += gsm0610_pack_voip(c, frame);
+            amp += GSM0610_FRAME_LEN;
+            break;
+        }
+        /*endswitch*/
+    }
+    /*endwhile*/
+    return (int) (c - code);
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/gsm0610_local.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/gsm0610_local.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,229 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * gsm0610_local.h - GSM 06.10 full rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the widely used GSM 06.10 code available from
+ * http://kbs.cs.tu-berlin.de/~jutta/toast.html
+ *
+ * $Id: gsm0610_local.h,v 1.7 2007/01/03 14:15:35 steveu Exp $
+ */
+
+#if !defined(_GSM0610_LOCAL_H_)
+#define _GSM0610_LOCAL_H_
+
+#define GSM0610_FRAME_LEN               160
+
+#define	GSM0610_MAGIC                   0xD
+
+static __inline__ int16_t gsm_add(int16_t a, int16_t b)
+{
+#if defined(__GNUC__)  &&  defined(__i386__)
+    __asm__ __volatile__(
+        " addw %2,%0;\n"
+        " jno 0f;\n"
+        " movw $0x7fff,%0;\n"
+        " adcw $0,%0;\n"
+        "0:"
+        : "=r" (a)
+        : "0" (a), "ir" (b)
+        : "cc"
+    );
+    return a;
+#else
+    int32_t sum;
+
+    sum = (int32_t) a + (int32_t) b;
+    return  saturate(sum);
+#endif
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int32_t gsm_l_add(int32_t a, int32_t b)
+{
+#if defined(__i386__)
+    __asm__ __volatile__(
+        " addl %2,%0;\n"
+        " jno 0f;\n"
+        " movl $0x7fffffff,%0;\n"
+        " adcl $0,%0;\n"
+        "0:"
+        : "=r" (a)
+        : "0" (a), "ir" (b)
+        : "cc"
+    );
+    return a;
+#else
+    uint32_t A;
+
+    if (a < 0)
+    {
+        if (b >= 0)
+            return  a + b;
+        /*endif*/
+        A = (uint32_t) -(a + 1) + (uint32_t) -(b + 1);
+        return  (A >= INT32_MAX)  ?  INT32_MIN  :  -(int32_t) A - 2;
+    }
+    /*endif*/
+    if (b <= 0)
+        return  a + b;
+    /*endif*/
+    A = (uint32_t) a + (uint32_t) b;
+    return  (A > INT32_MAX)  ?  INT32_MAX  :  A;
+#endif
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int16_t gsm_sub(int16_t a, int16_t b)
+{
+    int32_t diff;
+
+    diff = (int32_t) a - (int32_t) b;
+    return  saturate(diff);
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int16_t gsm_mult(int16_t a, int16_t b)
+{
+    if (a == INT16_MIN  &&  b == INT16_MIN)
+        return  INT16_MAX;
+    /*endif*/
+    return  (int16_t) (((int32_t) a * (int32_t) b) >> 15);
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int32_t gsm_l_mult(int16_t a, int16_t b)
+{
+    assert (a != INT16_MIN  ||  b != INT16_MIN);
+    return  ((int32_t) a * (int32_t) b) << 1;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int16_t gsm_mult_r(int16_t a, int16_t b)
+{
+    int32_t prod;
+
+    if (b == INT16_MIN  &&  a == INT16_MIN)
+        return  INT16_MAX;
+    /*endif*/
+    prod = (int32_t) a * (int32_t) b + 16384;
+    prod >>= 15;
+    return  (int16_t) (prod & 0xFFFF);
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int16_t gsm_abs(int16_t a)
+{
+    return  (a == INT16_MIN)  ?  INT16_MAX  :  (int16_t) abs(a);
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int16_t gsm_asr(int16_t a, int n)
+{
+    if (n >= 16)
+        return  -(a < 0);
+    /*endif*/
+    if (n <= -16)
+        return  0;
+    /*endif*/
+    if (n < 0)
+        return a << -n;
+    /*endif*/
+    return  a >> n;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int16_t gsm_asl(int16_t a, int n)
+{
+    if (n >= 16)
+        return  0;
+    /*endif*/
+    if (n <= -16)
+        return  -(a < 0);
+    /*endif*/
+    if (n < 0)
+        return  gsm_asr(a, -n);
+    /*endif*/
+    return  a << n;
+}
+/*- End of function --------------------------------------------------------*/
+
+extern void gsm0610_long_term_predictor(gsm0610_state_t *s,
+                                        int16_t d[40],
+                                        int16_t *dp,        /* [-120..-1] d'        IN  */
+                                        int16_t e[40],
+                                        int16_t dpp[40],
+                                        int16_t *Nc,
+                                        int16_t *bc);
+
+extern void gsm0610_lpc_analysis(gsm0610_state_t *s,
+                                 int16_t amp[160],
+                                 int16_t LARc[8]);
+
+extern void gsm0610_preprocess(gsm0610_state_t *s,
+                               const int16_t amp[],
+                               int16_t so[]);
+
+extern void gsm0610_short_term_analysis_filter(gsm0610_state_t *s,
+                                               int16_t LARc[8],
+                                               int16_t amp[160]);
+
+extern void gsm0610_long_term_synthesis_filtering(gsm0610_state_t *s,
+                                                  int16_t Ncr,
+                                                  int16_t bcr,
+                                                  int16_t erp[40],
+                                                  int16_t *drp);             /* [-120..-1] IN, [0..40] OUT */
+
+extern void gsm0610_rpe_decoding(gsm0610_state_t *s,
+                                 int16_t xmaxcr,
+                                 int16_t Mcr,
+                                 int16_t *xMcr,             /* [0..12], 3 bits             IN      */
+                                 int16_t erp[40]);
+
+extern void gsm0610_rpe_encoding(gsm0610_state_t *s,
+                                 int16_t *e,                /* [-5..-1][0..39][40..44]     IN/OUT  */
+                                 int16_t *xmaxc,
+                                 int16_t *Mc,
+                                 int16_t xMc[13]);
+
+extern void gsm0610_short_term_synthesis_filter(gsm0610_state_t *s,
+                                                int16_t LARcr[8],
+                                                int16_t drp[40],
+                                                int16_t amp[160]);
+
+extern int16_t gsm0610_norm(int32_t a);
+
+#if defined(__GNUC__)  &&  defined(__i386__)
+
+void gsm0610_vec_vsraw(const int16_t *p, int n, int bits);
+
+int32_t gsm0610_vec_iprod(const int16_t *p, const int16_t *q, int n);
+
+int32_t gsm0610_vec_maxmin(const int16_t *p, int n, int16_t *out);
+
+int32_t gsm0610_max_cross_corr(const int16_t *wt, const int16_t *dp, int16_t *Nc_out);
+
+#endif
+
+#endif
+
+/*- End of include ---------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/gsm0610_long_term.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/gsm0610_long_term.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,408 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * gsm0610_long_term.c - GSM 06.10 full rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the widely used GSM 06.10 code available from
+ * http://kbs.cs.tu-berlin.de/~jutta/toast.html
+ *
+ * $Id: gsm0610_long_term.c,v 1.10 2007/08/20 15:22:22 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <assert.h>
+#include <inttypes.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+#include <stdlib.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/bitstream.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/gsm0610.h"
+
+#include "gsm0610_local.h"
+
+/* Table 4.3a  Decision level of the LTP gain quantizer */
+static const int16_t gsm_DLB[4] =
+{
+    6554,    16384,    26214,     32767
+};
+
+/* Table 4.3b   Quantization levels of the LTP gain quantizer */
+static const int16_t gsm_QLB[4] =
+{
+    3277,    11469,    21299,     32767
+};
+
+/* 4.2.11 .. 4.2.12 LONG TERM PREDICTOR (LTP) SECTION */
+
+#if defined(__GNUC__)  &&  defined(__i386__)
+int32_t gsm0610_max_cross_corr(const int16_t *wt, const int16_t *dp, int16_t *Nc_out)
+{
+    int32_t lmax;
+    int32_t out;
+
+    __asm__ __volatile__(
+        " emms;\n"
+        " pushl %%ebx;\n"
+        " movl $0,%%edx;\n"             /* Will be maximum inner-product */
+        " movl $40,%%ebx;\n"
+        " movl %%ebx,%%ecx;\n"          /* Will be index of max inner-product */
+        " subl $80,%%esi;\n"
+        " .p2align 2;\n"
+        "1:\n"
+        " movq (%%edi),%%mm0;\n"
+        " movq (%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm0;\n"
+        " movq 8(%%edi),%%mm1;\n"
+        " movq 8(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq 16(%%edi),%%mm1;\n"
+        " movq 16(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq 24(%%edi),%%mm1;\n"
+        " movq 24(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq 32(%%edi),%%mm1;\n"
+        " movq 32(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq 40(%%edi),%%mm1;\n"
+        " movq 40(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq 48(%%edi),%%mm1;\n"
+        " movq 48(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq 56(%%edi),%%mm1;\n"
+        " movq 56(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq 64(%%edi),%%mm1;\n"
+        " movq 64(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq 72(%%edi),%%mm1;\n"
+        " movq 72(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq %%mm0,%%mm1;\n"
+        " punpckhdq %%mm0,%%mm1;\n"        /* mm1 has high int32 of mm0 dup'd */
+        " paddd %%mm1,%%mm0;\n"
+        " movd %%mm0,%%eax;\n"                /* eax has result */
+        " cmpl %%edx,%%eax;\n"
+        " jle 2f;\n"
+        " movl %%eax,%%edx;\n"
+        " movl %%ebx,%%ecx;\n"
+        " .p2align 2;\n"
+        "2:\n"
+        " subl $2,%%esi;\n"
+        " incl %%ebx;\n"
+        " cmpl $120,%%ebx;\n"
+        " jle 1b;\n"
+        " popl %%ebx;\n"
+        " emms;\n"
+        : "=d" (lmax), "=c" (out)
+        : "D" (wt), "S" (dp)
+        : "eax"
+    );
+    *Nc_out = out;
+    return  lmax;
+}
+/*- End of function --------------------------------------------------------*/
+#endif
+
+/* This procedure computes the LTP gain (bc) and the LTP lag (Nc)
+   for the long term analysis filter.   This is done by calculating a
+   maximum of the cross-correlation function between the current
+   sub-segment short term residual signal d[0..39] (output of
+   the short term analysis filter; for simplification the index
+   of this array begins at 0 and ends at 39 for each sub-segment of the
+   RPE-LTP analysis) and the previous reconstructed short term
+   residual signal dp[ -120 .. -1 ].  A dynamic scaling must be
+   performed to avoid overflow. */
+
+/* This procedure exists in three versions.  First, the integer
+   version; then, the two floating point versions (as another
+   function), with or without scaling. */
+
+static int16_t evaluate_ltp_parameters(int16_t d[40],
+                                       int16_t *dp,   // [-120..-1]   IN
+                                       int16_t *Nc_out)
+{
+    int k;
+    int16_t Nc;
+    int16_t bc;
+    int16_t wt[40];
+    int32_t L_max;
+    int32_t L_power;
+    int16_t R;
+    int16_t S;
+    int16_t dmax;
+    int16_t scale;
+    int16_t temp;
+    int32_t L_temp;
+#if !(defined(__GNUC__)  &&  defined(__i386__))
+    int16_t lambda;
+#endif
+
+    /* Search of the optimum scaling of d[0..39]. */
+    dmax = 0;
+    for (k = 0;  k < 40;  k++)
+    {
+        temp = d[k];
+        temp = gsm_abs(temp);
+        if (temp > dmax)
+            dmax = temp;
+        /*endif*/
+    }
+    /*endfor*/
+
+    if (dmax == 0)
+    {
+        temp = 0;
+    }
+    else
+    {
+        assert(dmax > 0);
+        temp = gsm0610_norm((int32_t) dmax << 16);
+    }
+    /*endif*/
+
+    if (temp > 6)
+        scale = 0;
+    else
+        scale = (int16_t) (6 - temp);
+    /*endif*/
+    assert(scale >= 0);
+
+    /* Initialization of a working array wt */
+    for (k = 0;  k < 40;  k++)
+        wt[k] = d[k] >> scale;
+    /*endfor*/
+
+    /* Search for the maximum cross-correlation and coding of the LTP lag */
+#if defined(__GNUC__)  &&  defined(__i386__)
+    L_max = gsm0610_max_cross_corr(wt, dp, &Nc);
+#else
+    L_max = 0;
+    Nc = 40; /* index for the maximum cross-correlation */
+
+    for (lambda = 40;  lambda <= 120;  lambda++)
+    {
+        int32_t L_result;
+
+        L_result  = (wt[0]*dp[0 - lambda])
+                  + (wt[1]*dp[1 - lambda])
+                  + (wt[2]*dp[2 - lambda])
+                  + (wt[3]*dp[3 - lambda])
+                  + (wt[4]*dp[4 - lambda])
+                  + (wt[5]*dp[5 - lambda])
+                  + (wt[6]*dp[6 - lambda])
+                  + (wt[7]*dp[7 - lambda])
+                  + (wt[8]*dp[8 - lambda])
+                  + (wt[9]*dp[9 - lambda])
+                  + (wt[10]*dp[10 - lambda])
+                  + (wt[11]*dp[11 - lambda])
+                  + (wt[12]*dp[12 - lambda])
+                  + (wt[13]*dp[13 - lambda])
+                  + (wt[14]*dp[14 - lambda])
+                  + (wt[15]*dp[15 - lambda])
+                  + (wt[16]*dp[16 - lambda])
+                  + (wt[17]*dp[17 - lambda])
+                  + (wt[18]*dp[18 - lambda])
+                  + (wt[19]*dp[19 - lambda])
+                  + (wt[20]*dp[20 - lambda])
+                  + (wt[21]*dp[21 - lambda])
+                  + (wt[22]*dp[22 - lambda])
+                  + (wt[23]*dp[23 - lambda])
+                  + (wt[24]*dp[24 - lambda])
+                  + (wt[25]*dp[25 - lambda])
+                  + (wt[26]*dp[26 - lambda])
+                  + (wt[27]*dp[27 - lambda])
+                  + (wt[28]*dp[28 - lambda])
+                  + (wt[29]*dp[29 - lambda])
+                  + (wt[30]*dp[30 - lambda])
+                  + (wt[31]*dp[31 - lambda])
+                  + (wt[32]*dp[32 - lambda])
+                  + (wt[33]*dp[33 - lambda])
+                  + (wt[34]*dp[34 - lambda])
+                  + (wt[35]*dp[35 - lambda])
+                  + (wt[36]*dp[36 - lambda])
+                  + (wt[37]*dp[37 - lambda])
+                  + (wt[38]*dp[38 - lambda])
+                  + (wt[39]*dp[39 - lambda]);
+
+        if (L_result > L_max)
+        {
+            Nc = lambda;
+            L_max = L_result;
+        }
+        /*endif*/
+    }
+    /*endfor*/
+#endif
+    *Nc_out = Nc;
+
+    L_max <<= 1;
+
+    /* Rescaling of L_max */
+    assert(scale <= 100  &&  scale >=  -100);
+    L_max = L_max >> (6 - scale);
+
+    assert(Nc <= 120  &&  Nc >= 40);
+
+    /* Compute the power of the reconstructed short term residual signal dp[..] */
+    L_power = 0;
+    for (k = 0;  k < 40;  k++)
+    {
+        L_temp = dp[k - Nc] >> 3;
+        L_power += L_temp*L_temp;
+    }
+    /*endfor*/
+    L_power <<= 1;  /* from L_MULT */
+
+    /* Normalization of L_max and L_power */
+    if (L_max <= 0)
+        return  0;
+    /*endif*/
+    if (L_max >= L_power)
+        return  3;
+    /*endif*/
+    temp = gsm0610_norm(L_power);
+
+    R = (int16_t) ((L_max << temp) >> 16);
+    S = (int16_t) ((L_power << temp) >> 16);
+
+    /* Coding of the LTP gain */
+
+    /* Table 4.3a must be used to obtain the level DLB[i] for the
+       quantization of the LTP gain b to get the coded version bc. */
+    for (bc = 0;  bc <= 2;  bc++)
+    {
+        if (R <= gsm_mult(S, gsm_DLB[bc]))
+            break;
+        /*endif*/
+    }
+    /*endfor*/
+    return  bc;
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.2.12 */
+static void long_term_analysis_filtering(int16_t bc,
+                                         int16_t Nc,
+                                         int16_t *dp,    // previous d   [-120..-1]      IN
+                                         int16_t d[40],
+                                         int16_t dpp[40],
+                                         int16_t e[40])
+{
+    int k;
+
+    /* In this part, we have to decode the bc parameter to compute
+       the samples of the estimate dpp[0..39].  The decoding of bc needs the
+       use of table 4.3b.  The long term residual signal e[0..39]
+       is then calculated to be fed to the RPE encoding section. */
+    for (k = 0;  k < 40;  k++)
+    {
+        dpp[k] = gsm_mult_r(gsm_QLB[bc], dp[k - Nc]);
+        e[k] = gsm_sub(d[k], dpp[k]);
+    }
+    /*endfor*/
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4x for 160 samples */
+void gsm0610_long_term_predictor(gsm0610_state_t *s,
+                                 int16_t d[40],
+                                 int16_t *dp,        // [-120..-1] d'    IN
+                                 int16_t e[40],
+                                 int16_t dpp[40],
+                                 int16_t *Nc,
+                                 int16_t *bc)
+{
+    assert(d);
+    assert(dp);
+    assert(e);
+    assert(dpp);
+    assert(Nc);
+    assert(bc);
+
+    *bc = evaluate_ltp_parameters(d, dp, Nc);
+    long_term_analysis_filtering(*bc, *Nc, dp, d, dpp, e);
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.3.2 */
+void gsm0610_long_term_synthesis_filtering(gsm0610_state_t *s,
+                                           int16_t Ncr,
+                                           int16_t bcr,
+                                           int16_t erp[40],
+                                           int16_t *drp)    // [-120..-1] IN, [0..40] OUT
+{
+    int k;
+    int16_t brp;
+    int16_t drpp;
+    int16_t Nr;
+
+    /* This procedure uses the bcr and Ncr parameter to realize the
+       long term synthesis filter.  The decoding of bcr needs
+       table 4.3b. */
+
+    /* Check the limits of Nr. */
+    Nr = (Ncr < 40  ||  Ncr > 120)  ?  s->nrp  :  Ncr;
+    s->nrp = Nr;
+    assert (Nr >= 40  &&  Nr <= 120);
+
+    /* Decode the LTP gain, bcr */
+    brp = gsm_QLB[bcr];
+
+    /* Compute the reconstructed short term residual signal, drp[0..39] */
+    assert(brp != INT16_MIN);
+    for (k = 0;  k < 40;  k++)
+    {
+        drpp = gsm_mult_r(brp, drp[k - Nr]);
+        drp[k] = gsm_add(erp[k], drpp);
+    }
+    /*endfor*/
+
+    /* Update the reconstructed short term residual signal, drp[-1..-120] */
+    for (k = 0;  k < 120;  k++)
+        drp[k - 120] = drp[k - 80];
+    /*endfor*/
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/gsm0610_lpc.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/gsm0610_lpc.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,533 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * gsm0610_lpc.c - GSM 06.10 full rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the widely used GSM 06.10 code available from
+ * http://kbs.cs.tu-berlin.de/~jutta/toast.html
+ *
+ * $Id: gsm0610_lpc.c,v 1.15 2007/08/20 15:22:22 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <assert.h>
+#include <inttypes.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+#include <stdlib.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/bitstream.h"
+#include "voipcodecs/bit_operations.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/vector_int.h"
+#include "voipcodecs/gsm0610.h"
+
+#include "gsm0610_local.h"
+
+/* 4.2.4 .. 4.2.7 LPC ANALYSIS SECTION */
+
+/* The number of left shifts needed to normalize the 32 bit
+   variable x for positive values on the interval
+   with minimum of
+   minimum of 1073741824  (01000000000000000000000000000000) and 
+   maximum of 2147483647  (01111111111111111111111111111111)
+   and for negative values on the interval with
+   minimum of -2147483648 (-10000000000000000000000000000000) and
+   maximum of -1073741824 ( -1000000000000000000000000000000).
+  
+   In order to normalize the result, the following
+   operation must be done: norm_var1 = x << gsm0610_norm(x);
+  
+   (That's 'ffs', only from the left, not the right..)
+*/
+
+int16_t gsm0610_norm(int32_t x)
+{
+    assert(x != 0);
+
+    if (x < 0)
+    {
+        if (x <= -1073741824)
+            return  0;
+        /*endif*/
+        x = ~x;
+    }
+    /*endif*/
+    return (int16_t) (30 - top_bit(x));
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+   (From p. 46, end of section 4.2.5)
+  
+   NOTE: The following lines gives [sic] one correct implementation
+         of the div(num, denum) arithmetic operation.  Compute div
+         which is the integer division of num by denom: with
+         denom >= num > 0
+*/
+static int16_t gsm_div(int16_t num, int16_t denom)
+{
+    int32_t num32;
+    int32_t denom32;
+    int16_t div;
+    int k;
+
+    /* The parameter num sometimes becomes zero.
+       Although this is explicitly guarded against in 4.2.5,
+       we assume that the result should then be zero as well. */
+
+    assert(num >= 0  &&  denom >= num);
+    if (num == 0)
+        return  0;
+    /*endif*/
+    num32 = num;
+    denom32 = denom;
+    div = 0;
+    k = 15;
+    while (k--)
+    {
+        div <<= 1;
+        num32 <<= 1;
+
+        if (num32 >= denom32)
+        {
+            num32 -= denom32;
+            div++;
+        }
+        /*endif*/
+    }
+    /*endwhile*/
+
+    return  div;
+}
+/*- End of function --------------------------------------------------------*/
+
+#if defined(__GNUC__)  &&  defined(__i386__)
+void gsm0610_vec_vsraw(const int16_t *p, int n, int bits)
+{
+    static const int64_t ones = 0x0001000100010001LL;
+
+    if (n == 0)
+        return;
+    /*endif*/
+    __asm__ __volatile__(
+        " leal -16(%%esi,%%eax,2),%%edx;\n"         /* edx = top - 16 */
+        " emms;\n"
+        " movd %%ecx,%%mm3;\n"
+        " movq %[ones],%%mm2;\n"
+        " psllw %%mm3,%%mm2;\n"
+        " psrlw $1,%%mm2;\n"
+        " cmpl %%edx,%%esi;"
+        " ja 4f;\n"
+
+         " .p2align 2;\n"
+        /* 8 words per iteration */
+        "6:\n"
+        " movq (%%esi),%%mm0;\n"
+        " movq 8(%%esi),%%mm1;\n"
+        " paddsw %%mm2,%%mm0;\n"
+        " psraw %%mm3,%%mm0;\n"
+        " paddsw %%mm2,%%mm1;\n"
+        " psraw %%mm3,%%mm1;\n"
+        " movq %%mm0,(%%esi);\n"
+        " movq %%mm1,8(%%esi);\n"
+        " addl $16,%%esi;\n"
+        " cmpl %%edx,%%esi;\n"
+        " jbe 6b;\n"
+
+        " .p2align 2;\n"
+        "4:\n"
+        " addl $12,%%edx;\n"                        /* now edx = top-4 */
+        " cmpl %%edx,%%esi;\n"
+        " ja 3f;\n"
+
+        " .p2align 2;\n"
+        /* do up to 6 words, two per iteration */
+        "5:\n"
+        " movd  (%%esi),%%mm0;\n"
+        " paddsw %%mm2,%%mm0;\n"
+        " psraw %%mm3,%%mm0;\n"
+        " movd %%mm0,(%%esi);\n"
+        " addl $4,%%esi;\n"
+        " cmpl %%edx,%%esi;\n"
+        " jbe 5b;\n"
+
+        " .p2align 2;\n"
+        "3:\n"
+        " addl $2,%%edx;\n"                        /* now edx = top-2 */
+        " cmpl %%edx,%%esi;\n"
+        " ja 2f;\n"
+        
+        " movzwl (%%esi),%%eax;\n"
+        " movd %%eax,%%mm0;\n"
+        " paddsw %%mm2,%%mm0;\n"
+        " psraw %%mm3,%%mm0;\n"
+        " movd %%mm0,%%eax;\n"
+        " movw %%ax,(%%esi);\n"
+
+        " .p2align 2;\n"
+        "2:\n"
+        " emms;\n"
+        :
+        : "S" (p), "a" (n), "c" (bits), [ones] "m" (ones)
+        : "edx"
+    );
+}
+/*- End of function --------------------------------------------------------*/
+#endif
+
+/* 4.2.4 */
+static void autocorrelation(int16_t amp[GSM0610_FRAME_LEN], int32_t L_ACF[9])
+{
+    int k;
+    int16_t smax;
+    int16_t scalauto;
+#if !(defined(__GNUC__)  &&  defined(__i386__))
+    int i;
+    int temp;
+    int16_t *sp;
+    int16_t sl;
+#endif
+    
+    /* The goal is to compute the array L_ACF[k].  The signal s[i] must
+       be scaled in order to avoid an overflow situation. */
+
+    /* Dynamic scaling of the array  s[0..159] */
+    /* Search for the maximum. */
+#if defined(__GNUC__)  &&  defined(__i386__)
+    smax = saturate(vec_min_maxi16(amp, GSM0610_FRAME_LEN, NULL));
+#else
+    for (smax = 0, k = 0;  k < GSM0610_FRAME_LEN;  k++)
+    {
+        temp = gsm_abs(amp[k]);
+        if (temp > smax)
+            smax = (int16_t) temp;
+        /*endif*/
+    }
+    /*endfor*/
+#endif
+
+    /* Computation of the scaling factor. */
+    if (smax == 0)
+    {
+        scalauto = 0;
+    }
+    else
+    {
+        assert(smax > 0);
+        scalauto = (int16_t) (4 - gsm0610_norm((int32_t) smax << 16));
+    }
+    /*endif*/
+
+    /* Scaling of the array s[0...159] */
+#if defined(__GNUC__)  &&  defined(__i386__)
+    if (scalauto > 0)
+        gsm0610_vec_vsraw(amp, GSM0610_FRAME_LEN, scalauto);
+    /*endif*/
+#else
+    if (scalauto > 0)
+    {
+        for (k = 0;  k < GSM0610_FRAME_LEN;  k++)
+            amp[k] = gsm_mult_r(amp[k], 16384 >> (scalauto - 1));
+        /*endfor*/
+    }
+    /*endif*/
+#endif
+
+    /* Compute the L_ACF[..]. */
+#if defined(__GNUC__)  &&  defined(__i386__)
+    for (k = 0;  k < 9;  k++)
+        L_ACF[k] = vec_dot_prodi16(amp, amp + k, GSM0610_FRAME_LEN - k) << 1;
+    /*endfor*/
+#else
+    sp = amp;
+    sl = *sp;
+    L_ACF[0] = ((int32_t) sl*sp[0]);
+    sl = *++sp;
+    L_ACF[0] += ((int32_t) sl*sp[0]);
+    L_ACF[1] = ((int32_t) sl*sp[-1]);
+    sl = *++sp;
+    L_ACF[0] += ((int32_t) sl*sp[0]);
+    L_ACF[1] += ((int32_t) sl*sp[-1]);
+    L_ACF[2] = ((int32_t) sl*sp[-2]);
+    sl = *++sp;
+    L_ACF[0] += ((int32_t) sl*sp[0]);
+    L_ACF[1] += ((int32_t) sl*sp[-1]);
+    L_ACF[2] += ((int32_t) sl*sp[-2]);
+    L_ACF[3] = ((int32_t) sl*sp[-3]);
+    sl = *++sp;
+    L_ACF[0] += ((int32_t) sl*sp[0]);
+    L_ACF[1] += ((int32_t) sl*sp[-1]);
+    L_ACF[2] += ((int32_t) sl*sp[-2]);
+    L_ACF[3] += ((int32_t) sl*sp[-3]);
+    L_ACF[4] = ((int32_t) sl*sp[-4]);
+    sl = *++sp;
+    L_ACF[0] += ((int32_t) sl*sp[0]);
+    L_ACF[1] += ((int32_t) sl*sp[-1]);
+    L_ACF[2] += ((int32_t) sl*sp[-2]);
+    L_ACF[3] += ((int32_t) sl*sp[-3]);
+    L_ACF[4] += ((int32_t) sl*sp[-4]);
+    L_ACF[5] = ((int32_t) sl*sp[-5]);
+    sl = *++sp;
+    L_ACF[0] += ((int32_t) sl*sp[0]);
+    L_ACF[1] += ((int32_t) sl*sp[-1]);
+    L_ACF[2] += ((int32_t) sl*sp[-2]);
+    L_ACF[3] += ((int32_t) sl*sp[-3]);
+    L_ACF[4] += ((int32_t) sl*sp[-4]);
+    L_ACF[5] += ((int32_t) sl*sp[-5]);
+    L_ACF[6] = ((int32_t) sl*sp[-6]);
+    sl = *++sp;
+    L_ACF[0] += ((int32_t) sl*sp[0]);
+    L_ACF[1] += ((int32_t) sl*sp[-1]);
+    L_ACF[2] += ((int32_t) sl*sp[-2]);
+    L_ACF[3] += ((int32_t) sl*sp[-3]);
+    L_ACF[4] += ((int32_t) sl*sp[-4]);
+    L_ACF[5] += ((int32_t) sl*sp[-5]);
+    L_ACF[6] += ((int32_t) sl*sp[-6]);
+    L_ACF[7] = ((int32_t) sl*sp[-7]);
+    sl = *++sp;
+    L_ACF[0] += ((int32_t) sl*sp[0]);
+    L_ACF[1] += ((int32_t) sl*sp[-1]);
+    L_ACF[2] += ((int32_t) sl*sp[-2]);
+    L_ACF[3] += ((int32_t) sl*sp[-3]);
+    L_ACF[4] += ((int32_t) sl*sp[-4]);
+    L_ACF[5] += ((int32_t) sl*sp[-5]);
+    L_ACF[6] += ((int32_t) sl*sp[-6]);
+    L_ACF[7] += ((int32_t) sl*sp[-7]);
+    L_ACF[8] = ((int32_t) sl*sp[-8]);
+    for (i = 9;  i < GSM0610_FRAME_LEN;  i++)
+    {
+        sl = *++sp;
+        L_ACF[0] += ((int32_t) sl*sp[0]);
+        L_ACF[1] += ((int32_t) sl*sp[-1]);
+        L_ACF[2] += ((int32_t) sl*sp[-2]);
+        L_ACF[3] += ((int32_t) sl*sp[-3]);
+        L_ACF[4] += ((int32_t) sl*sp[-4]);
+        L_ACF[5] += ((int32_t) sl*sp[-5]);
+        L_ACF[6] += ((int32_t) sl*sp[-6]);
+        L_ACF[7] += ((int32_t) sl*sp[-7]);
+        L_ACF[8] += ((int32_t) sl*sp[-8]);
+    }
+    /*endfor*/
+    for (k = 0;  k < 9;  k++)
+        L_ACF[k] <<= 1;
+    /*endfor*/
+#endif
+    /* Rescaling of the array s[0..159] */
+    if (scalauto > 0)
+    {
+        assert(scalauto <= 4); 
+        for (k = 0;  k < GSM0610_FRAME_LEN;  k++)
+            amp[k] <<= scalauto;
+        /*endfor*/
+    }
+    /*endif*/
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.2.5 */
+static void reflection_coefficients(int32_t L_ACF[9], int16_t r[8])
+{
+    int i;
+    int m;
+    int n;
+    int16_t temp;
+    int16_t ACF[9];
+    int16_t P[9];
+    int16_t K[9];
+
+    /* Schur recursion with 16 bits arithmetic. */
+    if (L_ACF[0] == 0)
+    {
+        for (i = 8;  i--;  *r++ = 0)
+            ;
+        /*endfor*/
+        return;
+    }
+    /*endif*/
+
+    assert(L_ACF[0] != 0);
+    temp = gsm0610_norm(L_ACF[0]);
+
+    assert(temp >= 0  &&  temp < 32);
+
+    /* ? overflow ? */
+    for (i = 0;  i <= 8;  i++)
+        ACF[i] = (int16_t) ((L_ACF[i] << temp) >> 16);
+    /*endfor*/
+
+    /* Initialize array P[..] and K[..] for the recursion. */
+    for (i = 1;  i <= 7;  i++)
+        K[i] = ACF[i];
+    /*endfor*/
+    for (i = 0;  i <= 8;  i++)
+        P[i] = ACF[i];
+    /*endfor*/
+    /* Compute reflection coefficients */
+    for (n = 1;  n <= 8;  n++, r++)
+    {
+        temp = P[1];
+        temp = gsm_abs (temp);
+        if (P[0] < temp)
+        {
+            for (i = n;  i <= 8;  i++)
+                *r++ = 0;
+            /*endfor*/
+            return;
+        }
+        /*endif*/
+
+        *r = gsm_div(temp, P[0]);
+
+        assert(*r >= 0);
+        if (P[1] > 0)
+            *r = -*r;     /* r[n] = sub(0, r[n]) */
+        /*endif*/
+        assert(*r != INT16_MIN);
+        if (n == 8)
+            return; 
+        /*endif*/
+
+        /* Schur recursion */
+        temp = gsm_mult_r(P[1], *r);
+        P[0] = gsm_add(P[0], temp);
+
+        for (m = 1;  m <= 8 - n;  m++)
+        {
+            temp = gsm_mult_r(K[m], *r);
+            P[m] = gsm_add(P[m + 1], temp);
+
+            temp = gsm_mult_r(P[m + 1], *r);
+            K[m] = gsm_add(K[m], temp);
+        }
+        /*endfor*/
+    }
+    /*endfor*/
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.2.6 */
+static void transform_to_log_area_ratios(int16_t r[8])
+{
+    int16_t temp;
+    int i;
+
+    /* The following scaling for r[..] and LAR[..] has been used:
+      
+       r[..]   = integer (real_r[..]*32768.); -1 <= real_r < 1.
+       LAR[..] = integer (real_LAR[..] * 16384);
+       with -1.625 <= real_LAR <= 1.625
+    */
+
+    /* Computation of the LAR[0..7] from the r[0..7] */
+    for (i = 1;  i <= 8;  i++, r++)
+    {
+        temp = *r;
+        temp = gsm_abs(temp);
+        assert(temp >= 0);
+
+        if (temp < 22118)
+        {
+            temp >>= 1;
+        }
+        else if (temp < 31130)
+        {
+            assert(temp >= 11059);
+            temp -= 11059;
+        }
+        else
+        {
+            assert(temp >= 26112);
+            temp -= 26112;
+            temp <<= 2;
+        }
+        /*endif*/
+
+        *r = (*r < 0)  ?  -temp  :  temp;
+        assert(*r != INT16_MIN);
+    }
+    /*endfor*/
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.2.7 */
+static void quantization_and_coding(int16_t LAR[8])
+{
+    int16_t temp;
+
+    /* This procedure needs four tables; the following equations
+       give the optimum scaling for the constants:
+        
+       A[0..7] = integer(real_A[0..7] * 1024)
+       B[0..7] = integer(real_B[0..7] *  512)
+       MAC[0..7] = maximum of the LARc[0..7]
+       MIC[0..7] = minimum of the LARc[0..7] */
+
+#undef STEP
+#define STEP(A,B,MAC,MIC)                                       \
+        temp = gsm_mult(A, *LAR);                               \
+        temp = gsm_add(temp, B);                                \
+        temp = gsm_add(temp, 256);                              \
+        temp >>= 9;                                             \
+        *LAR  = (int16_t) ((temp > MAC)                         \
+                         ?                                      \
+                         MAC - MIC                              \
+                         :                                      \
+                         ((temp < MIC)  ?  0  :  temp - MIC));  \
+        LAR++;
+
+    STEP(20480,     0,  31, -32);
+    STEP(20480,     0,  31, -32);
+    STEP(20480,  2048,  15, -16);
+    STEP(20480, -2560,  15, -16);
+
+    STEP(13964,    94,   7,  -8);
+    STEP(15360, -1792,   7,  -8);
+    STEP( 8534,  -341,   3,  -4);
+    STEP( 9036, -1144,   3,  -4);
+#undef STEP
+}
+/*- End of function --------------------------------------------------------*/
+
+void gsm0610_lpc_analysis(gsm0610_state_t *s,
+                          int16_t amp[GSM0610_FRAME_LEN],
+                          int16_t LARc[8])
+{
+    int32_t L_ACF[9];
+
+    autocorrelation(amp, L_ACF);
+    reflection_coefficients(L_ACF, LARc);
+    transform_to_log_area_ratios(LARc);
+    quantization_and_coding(LARc);
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/gsm0610_preprocess.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/gsm0610_preprocess.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,149 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * gsm0610_preprocess.c - GSM 06.10 full rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the widely used GSM 06.10 code available from
+ * http://kbs.cs.tu-berlin.de/~jutta/toast.html
+ *
+ * $Id: gsm0610_preprocess.c,v 1.8 2007/08/20 15:22:22 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <assert.h>
+#include <inttypes.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+#include <stdlib.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/bitstream.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/gsm0610.h"
+
+#include "gsm0610_local.h"
+
+/*
+    4.2.0 .. 4.2.3  PREPROCESSING SECTION
+    
+    After A-law to linear conversion (or directly from the
+    A to D converter) the following scaling is assumed for
+    input to the RPE-LTP algorithm:
+  
+    in:  0.1.....................12
+         S.v.v.v.v.v.v.v.v.v.v.v.v.*.*.*
+  
+    Where S is the sign bit, v a valid bit, and * a "don't care" bit.
+    The original signal is called sop[..]
+  
+    out:   0.1................... 12 
+         S.S.v.v.v.v.v.v.v.v.v.v.v.v.0.0
+*/
+
+void gsm0610_preprocess(gsm0610_state_t *s, const int16_t amp[GSM0610_FRAME_LEN], int16_t so[GSM0610_FRAME_LEN])
+{
+    int16_t z1;
+    int16_t mp;
+    int16_t s1;
+    int16_t msp;
+    int16_t SO;
+    int32_t L_z2;
+    int32_t L_s2;
+    int32_t L_temp;
+#if !defined(__GNUC__)
+    int16_t lsp;
+#endif
+    int k;
+
+    z1 = s->z1;
+    L_z2 = s->L_z2;
+    mp = s->mp;
+    for (k = 0;  k < GSM0610_FRAME_LEN;  k++)
+    {
+        /* 4.2.1   Downscaling of the input signal */
+        SO = (amp[k] >> 1) & ~3;
+
+        assert(SO >= -0x4000); // downscaled by
+        assert(SO <=  0x3FFC); // previous routine.
+
+        /* 4.2.2   Offset compensation */
+
+        /*  This part implements a high-pass filter and requires extended
+            arithmetic precision for the recursive part of this filter.
+            The input of this procedure is the array so[0...159] and the
+            output the array sof[ 0...159 ].
+        */
+        /* Compute the non-recursive part */
+        s1 = SO - z1;
+        z1 = SO;
+
+        assert(s1 != INT16_MIN);
+
+        /* Compute the recursive part */
+        L_s2 = s1;
+        L_s2 <<= 15;
+
+        /* Perform a 31 by 16 bits multiplication */
+#if defined(__GNUC__)
+        L_z2 = ((int64_t) L_z2*32735 + 0x4000) >> 15;
+        /* Alternate (ANSI) version of below line does slightly different rounding:
+         * L_temp = L_z2 >> 9;
+         * L_temp += L_temp >> 5;
+         * L_temp = (++L_temp) >> 1;
+         * L_z2 = L_z2 - L_temp;
+         */
+        L_z2 = gsm_l_add(L_z2, L_s2);
+#else
+        /* This does L_z2  = L_z2 * 0x7FD5/0x8000 + L_s2 */
+        msp = (int16_t) (L_z2 >> 15);
+        lsp = (int16_t) (L_z2 - ((int32_t) msp << 15));
+
+        L_s2 += gsm_mult_r(lsp, 32735);
+        L_temp = (int32_t) msp*32735;
+        L_z2 = gsm_l_add(L_temp, L_s2);
+#endif
+
+        /* Compute sof[k] with rounding */
+        L_temp = gsm_l_add(L_z2, 16384);
+
+        /* 4.2.3  Preemphasis */
+        msp = gsm_mult_r(mp, -28180);
+        mp = (int16_t) (L_temp >> 15);
+        so[k] = gsm_add(mp, msp);
+    }
+    /*endfor*/
+
+    s->z1 = z1;
+    s->L_z2 = L_z2;
+    s->mp = mp;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/gsm0610_rpe.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/gsm0610_rpe.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,531 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * gsm0610_rpe.c - GSM 06.10 full rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the widely used GSM 06.10 code available from
+ * http://kbs.cs.tu-berlin.de/~jutta/toast.html
+ *
+ * $Id: gsm0610_rpe.c,v 1.14 2007/08/20 15:22:22 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <assert.h>
+#include <inttypes.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+#include <stdlib.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/bitstream.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/gsm0610.h"
+
+#include "gsm0610_local.h"
+
+/* 4.2.13 .. 4.2.17  RPE ENCODING SECTION */
+
+/* 4.2.13 */
+static void weighting_filter(const int16_t *e,     // signal [-5..0.39.44] IN
+                             int16_t x[40])
+{
+#if defined(__GNUC__)  &&  defined(__i386__)
+    /* Table 4.4   Coefficients of the weighting filter */
+    /* This must be padded to a multiple of 4 for MMX to work */
+    static const int16_t gsm_H[12] =
+    {
+        -134, -374, 0, 2054, 5741, 8192, 5741, 2054, 0, -374, -134, 0
+    };
+
+    __asm__ __volatile__(
+        " emms;\n"
+        " addl $-10,%%ecx;\n"
+        " movl $0x1000,%%eax;\n"
+        " movd %%eax,%%mm5;\n"              /* for rounding */
+        " movq %[gsm_H],%%mm1;\n"
+        " movq %[gsm_H8],%%mm2;\n"
+        " movq %[gsm_H16],%%mm3;\n"
+        " xorl %%esi,%%esi;\n"
+        " .p2align 2;\n"
+        "1:;\n"
+        " movq (%%ecx,%%esi,2),%%mm0;\n"
+        " pmaddwd %%mm1,%%mm0;\n"
+ 
+        " movq 8(%%ecx,%%esi,2),%%mm4;\n"
+        " pmaddwd %%mm2,%%mm4;\n"
+        " paddd %%mm4,%%mm0;\n"
+
+        " movq 16(%%ecx,%%esi,2),%%mm4;\n"
+        " pmaddwd %%mm3,%%mm4;\n"
+        " paddd %%mm4,%%mm0;\n"
+
+        " movq %%mm0,%%mm4;\n"
+        " punpckhdq %%mm0,%%mm4;\n"         /* mm4 has high int32 of mm0 dup'd */
+        " paddd %%mm4,%%mm0;\n"
+
+        " paddd %%mm5,%%mm0;\n"             /* Add for roundoff */
+        " psrad $13,%%mm0;\n"
+        " packssdw %%mm0,%%mm0;\n"        
+        " movd %%mm0,%%eax;\n"              /* eax has result */
+        " movw %%ax,(%%edi,%%esi,2);\n"
+        " incl %%esi;\n"
+        " cmpl $39,%%esi;\n"
+        " jle 1b;\n"
+        " emms;\n"
+        :
+        : "c" (e), "D" (x), [gsm_H] "X" (*((int64_t *) gsm_H)), [gsm_H8] "X" (*((int64_t *) (gsm_H + 4))), [gsm_H16] "X" (*((int64_t *) (gsm_H + 8)))
+        : "eax", "edx", "esi"
+    );
+#else
+    int32_t L_result;
+    int k;
+
+    /* The coefficients of the weighting filter are stored in a table
+       (see table 4.4).  The following scaling is used:
+      
+        H[0..10] = integer(real_H[0..10] * 8192); 
+    */
+    /* Initialization of a temporary working array wt[0...49] */
+
+    /* for (k =  0;  k <=  4;  k++) wt[k] = 0;
+     * for (k =  5;  k <= 44;  k++) wt[k] = *e++;
+     * for (k = 45;  k <= 49;  k++) wt[k] = 0;
+     *
+     *  (e[-5..-1] and e[40..44] are allocated by the caller,
+     *  are initially zero and are not written anywhere.)
+     */
+    e -= 5;
+
+    /* Compute the signal x[0..39] */
+    for (k = 0;  k < 40;  k++)
+    {
+        L_result = 8192 >> 1;
+
+        /* for (i = 0; i <= 10; i++)
+         * {
+         *      L_temp   = gsm_l_mult(wt[k + i], gsm_H[i]);
+         *      L_result = gsm_l_add(L_result, L_temp);
+         * }
+         */
+
+#undef STEP
+#define STEP(i,H) (e[k + i] * (int32_t) H)
+
+        /* Every one of these multiplications is done twice,
+           but I don't see an elegant way to optimize this. 
+           Do you?
+        */
+        L_result += STEP( 0,  -134);
+        L_result += STEP( 1,  -374);
+              // += STEP( 2,  0   );
+        L_result += STEP( 3,  2054);
+        L_result += STEP( 4,  5741);
+        L_result += STEP( 5,  8192);
+        L_result += STEP( 6,  5741);
+        L_result += STEP( 7,  2054);
+              // += STEP( 8,  0   );
+        L_result += STEP( 9,  -374);
+        L_result += STEP(10,  -134);
+
+        /* 2 adds vs. >> 16 => 14, minus one shift to compensate for
+           those we lost when replacing L_MULT by '*'. */
+        L_result >>= 13;
+        x[k] = saturate(L_result);
+    }
+    /*endfor*/
+#endif
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.2.14 */
+static void rpe_grid_selection(int16_t x[40], int16_t xM[13], int16_t *Mc_out)
+{
+    int i;
+    int32_t L_result;
+    int32_t L_temp;
+    int32_t EM;     /* xxx should be L_EM? */
+    int16_t Mc;
+    int32_t L_common_0_3;
+
+    /* The signal x[0..39] is used to select the RPE grid which is
+       represented by Mc. */
+
+    EM = 0;
+    Mc = 0;
+
+#undef STEP
+#define STEP(m,i)                           \
+    L_temp = x[m + 3*i] >> 2;               \
+    L_result += L_temp*L_temp;
+
+    /* Common part of 0 and 3 */
+    L_result = 0;
+    STEP(0, 1);
+    STEP(0, 2);
+    STEP(0, 3);
+    STEP(0, 4);
+    STEP(0, 5);
+    STEP(0, 6);
+    STEP(0, 7);
+    STEP(0, 8);
+    STEP(0, 9);
+    STEP(0, 10);
+    STEP(0, 11);
+    STEP(0, 12);
+    L_common_0_3 = L_result;
+
+    /* i = 0 */
+
+    STEP(0, 0);
+    L_result <<= 1; /* implicit in L_MULT */
+    EM = L_result;
+
+    /* i = 1 */
+
+    L_result = 0;
+    STEP(1, 0);
+    STEP(1, 1);
+    STEP(1, 2);
+    STEP(1, 3);
+    STEP(1, 4);
+    STEP(1, 5);
+    STEP(1, 6);
+    STEP(1, 7);
+    STEP(1, 8);
+    STEP(1, 9);
+    STEP(1, 10);
+    STEP(1, 11);
+    STEP(1, 12);
+    L_result <<= 1;
+    if (L_result > EM)
+    {
+        Mc = 1;
+        EM = L_result;
+    }
+    /*endif*/
+
+    /* i = 2 */
+
+    L_result = 0;
+    STEP(2, 0);
+    STEP(2, 1);
+    STEP(2, 2);
+    STEP(2, 3);
+    STEP(2, 4);
+    STEP(2, 5);
+    STEP(2, 6);
+    STEP(2, 7);
+    STEP(2, 8);
+    STEP(2, 9);
+    STEP(2, 10);
+    STEP(2, 11);
+    STEP(2, 12);
+    L_result <<= 1;
+    if (L_result > EM)
+    {
+        Mc = 2;
+        EM = L_result;
+    }
+    /*endif*/
+
+    /* i = 3 */
+
+    L_result = L_common_0_3;
+    STEP(3, 12);
+    L_result <<= 1;
+    if (L_result > EM)
+    {
+        Mc = 3;
+        EM = L_result;
+    }
+    /*endif*/
+
+    /* Down-sampling by a factor 3 to get the selected xM[0..12]
+       RPE sequence. */
+    for (i = 0;  i < 13;  i++)
+        xM[i] = x[Mc + 3*i];
+    /*endfor*/
+    *Mc_out = Mc;
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.12.15 */
+static void apcm_quantization_xmaxc_to_exp_mant(int16_t xmaxc,
+                                                int16_t *exp_out,
+                                                int16_t *mant_out)
+{
+    int16_t exp;
+    int16_t mant;
+
+    /* Compute exponent and mantissa of the decoded version of xmaxc */
+    exp = 0;
+    if (xmaxc > 15)
+        exp = (int16_t) ((xmaxc >> 3) - 1);
+    /*endif*/
+    mant = xmaxc - (exp << 3);
+
+    if (mant == 0)
+    {
+        exp = -4;
+        mant = 7;
+    }
+    else
+    {
+        while (mant <= 7)
+        {
+            mant = (int16_t) (mant << 1 | 1);
+            exp--;
+        }
+        /*endwhile*/
+        mant -= 8;
+    }
+    /*endif*/
+
+    assert(exp >= -4  &&  exp <= 6);
+    assert(mant >= 0  &&  mant <= 7);
+
+    *exp_out  = exp;
+    *mant_out = mant;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void apcm_quantization(int16_t xM[13],
+                              int16_t xMc[13],
+                              int16_t *mant_out,
+                              int16_t *exp_out,
+                              int16_t *xmaxc_out)
+{
+    /* Table 4.5   Normalized inverse mantissa used to compute xM/xmax */
+    static const int16_t gsm_NRFAC[8] =
+    {
+        29128, 26215, 23832, 21846, 20165, 18725, 17476, 16384
+    };
+    int i;
+    int itest;
+    int16_t xmax;
+    int16_t xmaxc;
+    int16_t temp;
+    int16_t temp1;
+    int16_t temp2;
+    int16_t exp;
+    int16_t mant;
+
+    /* Find the maximum absolute value xmax of xM[0..12]. */
+    xmax = 0;
+    for (i = 0;  i < 13;  i++)
+    {
+        temp = xM[i];
+        temp = gsm_abs(temp);
+        if (temp > xmax)
+            xmax = temp;
+        /*endif*/
+    }
+    /*endfor*/
+
+    /* Quantizing and coding of xmax to get xmaxc. */
+    exp = 0;
+    temp = xmax >> 9;
+    itest = 0;
+
+    for (i = 0;  i <= 5;  i++)
+    {
+        itest |= (temp <= 0);
+        temp >>= 1;
+
+        assert(exp <= 5);
+        if (itest == 0)
+            exp++;
+        /*endif*/
+    }
+    /*endfor*/
+
+    assert(exp <= 6  &&  exp >= 0);
+    temp = (int16_t) (exp + 5);
+
+    assert(temp <= 11  &&  temp >= 0);
+    xmaxc = gsm_add((xmax >> temp), exp << 3);
+
+    /* Quantizing and coding of the xM[0..12] RPE sequence
+       to get the xMc[0..12] */
+    apcm_quantization_xmaxc_to_exp_mant(xmaxc, &exp, &mant);
+
+    /* This computation uses the fact that the decoded version of xmaxc
+       can be calculated by using the exponent and the mantissa part of
+       xmaxc (logarithmic table).
+       So, this method avoids any division and uses only a scaling
+       of the RPE samples by a function of the exponent.  A direct 
+       multiplication by the inverse of the mantissa (NRFAC[0..7]
+       found in table 4.5) gives the 3 bit coded version xMc[0..12]
+       of the RPE samples.
+    */
+    /* Direct computation of xMc[0..12] using table 4.5 */
+    assert(exp <= 4096  &&  exp >= -4096);
+    assert(mant >= 0  &&  mant <= 7); 
+
+    temp1 = (int16_t) (6 - exp);    /* Normalization by the exponent */
+    temp2 = gsm_NRFAC[mant];        /* Inverse mantissa */
+
+    for (i = 0;  i < 13;  i++)
+    {
+        assert(temp1 >= 0  &&  temp1 < 16);
+
+        temp = xM[i] << temp1;
+        temp = gsm_mult(temp, temp2);
+        temp >>= 12;
+        xMc[i] = (int16_t) (temp + 4);      /* See note below */
+    }
+    /*endfor*/
+
+    /* NOTE: This equation is used to make all the xMc[i] positive. */
+    *mant_out  = mant;
+    *exp_out   = exp;
+    *xmaxc_out = xmaxc;
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.2.16 */
+static void apcm_inverse_quantization(int16_t xMc[13],
+                                      int16_t mant,
+                                      int16_t exp,
+                                      int16_t xMp[13])
+{
+    /* Table 4.6   Normalized direct mantissa used to compute xM/xmax */
+    static const int16_t gsm_FAC[8] =
+    {
+        18431, 20479, 22527, 24575, 26623, 28671, 30719, 32767
+    };
+    int i;
+    int16_t temp;
+    int16_t temp1;
+    int16_t temp2;
+    int16_t temp3;
+
+    /* This part is for decoding the RPE sequence of coded xMc[0..12]
+       samples to obtain the xMp[0..12] array.  Table 4.6 is used to get
+       the mantissa of xmaxc (FAC[0..7]).
+    */
+    assert(mant >= 0  &&  mant <= 7);
+
+    temp1 = gsm_FAC[mant];          /* See 4.2-15 for mant */
+    temp2 = gsm_sub(6, exp);        /* See 4.2-15 for exp */
+    temp3 = gsm_asl(1, gsm_sub (temp2, 1));
+
+    for (i = 0;  i < 13;  i++)
+    {
+        assert(xMc[i] >= 0  &&  xMc[i] <= 7);   /* 3 bit unsigned */
+
+        temp = (int16_t) ((xMc[i] << 1) - 7);   /* Restore sign */
+        assert(temp <= 7  &&  temp >= -7);      /* 4 bit signed */
+
+        temp <<= 12;                            /* 16 bit signed */
+        temp = gsm_mult_r(temp1, temp);
+        temp = gsm_add(temp, temp3);
+        xMp[i] = gsm_asr(temp, temp2);
+    }
+    /*endfor*/
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.2.17 */
+static void rpe_grid_positioning(int16_t Mc,
+                                 int16_t xMp[13],
+                                 int16_t ep[40])
+{
+    int i = 13;
+
+    /* This procedure computes the reconstructed long term residual signal
+       ep[0..39] for the LTP analysis filter.  The inputs are the Mc
+       which is the grid position selection and the xMp[0..12] decoded
+       RPE samples which are upsampled by a factor of 3 by inserting zero
+       values.
+    */
+    assert(0 <= Mc  &&  Mc <= 3);
+
+    switch (Mc)
+    {
+    case 3:
+        *ep++ = 0;
+    case 2:
+        do
+        {
+            *ep++ = 0;
+    case 1:
+            *ep++ = 0;
+    case 0:
+            *ep++ = *xMp++;
+        }
+        while (--i);
+    }
+    /*endswitch*/
+    while (++Mc < 4)
+        *ep++ = 0;
+    /*endwhile*/
+}
+/*- End of function --------------------------------------------------------*/
+
+void gsm0610_rpe_encoding(gsm0610_state_t *s,
+                          int16_t *e,          // [-5..-1][0..39][40..44]
+                          int16_t *xmaxc,
+                          int16_t *Mc,
+                          int16_t xMc[13])
+{
+    int16_t x[40];
+    int16_t xM[13];
+    int16_t xMp[13];
+    int16_t mant;
+    int16_t exp;
+
+    weighting_filter(e, x);
+    rpe_grid_selection(x, xM, Mc);
+
+    apcm_quantization(xM, xMc, &mant, &exp, xmaxc);
+    apcm_inverse_quantization(xMc, mant, exp, xMp);
+
+    rpe_grid_positioning(*Mc, xMp, e);
+}
+/*- End of function --------------------------------------------------------*/
+
+void gsm0610_rpe_decoding(gsm0610_state_t *s,
+                          int16_t xmaxc,
+                          int16_t Mcr,
+                          int16_t xMcr[13],
+                          int16_t erp[40])
+{
+    int16_t exp;
+    int16_t mant;
+    int16_t xMp[13];
+
+    apcm_quantization_xmaxc_to_exp_mant(xmaxc, &exp, &mant);
+    apcm_inverse_quantization(xMcr, mant, exp, xMp);
+    rpe_grid_positioning(Mcr, xMp, erp);
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/gsm0610_short_term.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/gsm0610_short_term.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,361 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * gsm0610_short_term.c - GSM 06.10 full rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the widely used GSM 06.10 code available from
+ * http://kbs.cs.tu-berlin.de/~jutta/toast.html
+ *
+ * $Id: gsm0610_short_term.c,v 1.10 2007/08/20 15:22:22 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <assert.h>
+#include <inttypes.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+#include <stdlib.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/bitstream.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/gsm0610.h"
+
+#include "gsm0610_local.h"
+
+/* SHORT TERM ANALYSIS FILTERING SECTION */
+
+/* 4.2.8 */
+static void decode_log_area_ratios(int16_t LARc[8], int16_t *LARpp)
+{
+    int16_t temp1;
+
+    /* This procedure requires for efficient implementation
+       two tables.
+       INVA[1..8] = integer((32768*8)/real_A[1..8])
+       MIC[1..8]  = minimum value of the LARc[1..8]
+    */
+
+    /* Compute the LARpp[1..8] */
+
+#undef STEP
+#define STEP(B,MIC,INVA)                        \
+    temp1 = gsm_add(*LARc++, MIC) << 10;       \
+    temp1 = gsm_sub(temp1, B << 1);            \
+    temp1 = gsm_mult_r (INVA, temp1);           \
+    *LARpp++ = gsm_add(temp1, temp1);
+
+    STEP(    0,  -32,  13107);
+    STEP(    0,  -32,  13107);
+    STEP( 2048,  -16,  13107);
+    STEP(-2560,  -16,  13107);
+
+    STEP(   94,   -8,  19223);
+    STEP(-1792,   -8,  17476);
+    STEP( -341,   -4,  31454);
+    STEP(-1144,   -4,  29708);
+
+    /* NOTE: the addition of *MIC is used to restore the sign of *LARc. */
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.2.9 */
+
+/* Computation of the quantized reflection coefficients */
+
+/* 4.2.9.1  Interpolation of the LARpp[1..8] to get the LARp[1..8] */
+
+/* Within each frame of 160 analyzed speech samples the short term
+   analysis and synthesis filters operate with four different sets of
+   coefficients, derived from the previous set of decoded LARs(LARpp(j - 1))
+   and the actual set of decoded LARs (LARpp(j))
+  
+   (Initial value: LARpp(j - 1)[1..8] = 0.)
+*/
+
+static void coefficients_0_12(int16_t *LARpp_j_1,
+                              int16_t *LARpp_j,
+                              int16_t *LARp)
+{
+    int i;
+
+    for (i = 1;  i <= 8;  i++, LARp++, LARpp_j_1++, LARpp_j++)
+    {
+        *LARp = gsm_add(*LARpp_j_1 >> 2, *LARpp_j >> 2);
+        *LARp = gsm_add(*LARp, *LARpp_j_1 >> 1);
+    }
+    /*endfor*/
+}
+/*- End of function --------------------------------------------------------*/
+
+static void coefficients_13_26(int16_t *LARpp_j_1,
+                               int16_t *LARpp_j,
+                               int16_t *LARp)
+{
+    int i;
+
+    for (i = 1;  i <= 8;  i++, LARpp_j_1++, LARpp_j++, LARp++)
+        *LARp = gsm_add(*LARpp_j_1 >> 1, *LARpp_j >> 1);
+    /*endfor*/
+}
+/*- End of function --------------------------------------------------------*/
+
+static void coefficients_27_39(int16_t *LARpp_j_1,
+                               int16_t *LARpp_j,
+                               int16_t *LARp)
+{
+    int i;
+
+    for (i = 1;  i <= 8;  i++, LARpp_j_1++, LARpp_j++, LARp++)
+    {
+        *LARp = gsm_add(*LARpp_j_1 >> 2, *LARpp_j >> 2);
+        *LARp = gsm_add(*LARp, *LARpp_j >> 1);
+    }
+    /*endfor*/
+}
+/*- End of function --------------------------------------------------------*/
+
+static void coefficients_40_159(int16_t *LARpp_j, int16_t *LARp)
+{
+    int i;
+
+    for (i = 1;  i <= 8;  i++)
+        *LARp++ = *LARpp_j++;
+    /*endfor*/
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.2.9.2 */
+static void larp_to_rp(int16_t LARp[8])
+{
+    int i;
+    int16_t *LARpx;
+    int16_t temp;
+
+    /* The input to this procedure is the interpolated LARp[0..7] array.
+       The reflection coefficients, rp[i], are used in the analysis
+       filter and in the synthesis filter.
+    */
+
+    LARpx = LARp;
+    for (i = 1;  i <= 8;  i++, LARpx++)
+    {
+        temp = *LARpx;
+        if (temp < 0)
+        {
+            if (temp == INT16_MIN)
+                temp = INT16_MAX;
+            else
+                temp = -temp;
+            /*endif*/
+            if (temp < 11059)
+                temp <<= 1;
+            else if (temp < 20070)
+                temp += 11059;
+            else
+                temp = gsm_add(temp >> 2, 26112);
+            /*endif*/
+            *LARpx = -temp;
+        }
+        else
+        {
+            if (temp < 11059)
+                temp <<= 1;
+            else if (temp < 20070)
+                temp += 11059;
+            else
+                temp = gsm_add(temp >> 2, 26112);
+            /*endif*/
+            *LARpx = temp;
+        }
+        /*endif*/
+    }
+    /*endfor*/
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 4.2.10 */
+static void short_term_analysis_filtering(gsm0610_state_t *s,
+                                          int16_t rp[8],
+                                          int k_n,          // k_end - k_start
+                                          int16_t amp[])    // [0..n-1]     IN/OUT
+{
+    /* This procedure computes the short term residual signal d[..] to be fed
+       to the RPE-LTP loop from the s[..] signal and from the local rp[..]
+       array (quantized reflection coefficients).  As the call of this
+       procedure can be done in many ways (see the interpolation of the LAR
+       coefficient), it is assumed that the computation begins with index
+       k_start (for arrays d[..] and s[..]) and stops with index k_end
+       (k_start and k_end are defined in 4.2.9.1).  This procedure also
+       needs to keep the array u[0..7] in memory for each call.
+    */
+    int16_t *u0;
+    int16_t *u_top;
+    int i;
+    int16_t *u;
+    int16_t *rpx;
+    int32_t di;
+    int32_t u_out;
+
+    u0 = s->u;
+    u_top = u0 + 8;
+
+    for (i = 0;  i < k_n;  i++)
+    {
+        di =
+        u_out = amp[i];
+        for (rpx = rp, u = u0;  u < u_top;  )
+        {
+            int32_t ui;
+            int32_t rpi;
+
+            ui = *u;
+            *u++ = (int16_t) u_out;
+            rpi = *rpx++;
+            u_out = ui + (((rpi*di) + 0x4000) >> 15);
+            di = di + (((rpi*ui) + 0x4000) >> 15);
+            u_out = saturate(u_out);
+            di = saturate(di);
+        }
+        /*endfor*/
+        amp[i] = (int16_t) di;
+    }
+    /*endfor*/
+}
+/*- End of function --------------------------------------------------------*/
+
+static void short_term_synthesis_filtering(gsm0610_state_t *s,
+                                           int16_t rrp[8],
+                                           int k,          // k_end - k_start
+                                           int16_t *wt,    // [0..k - 1]
+                                           int16_t *sr)    // [0..k - 1]
+{
+    int16_t *v;
+    int i;
+    int16_t sri;
+    int16_t tmp1;
+    int16_t tmp2;
+
+    v = s->v;
+    while (k--)
+    {
+        sri = *wt++;
+        for (i = 8;  i--;  )
+        {
+            tmp1 = rrp[i];
+            tmp2 = v[i];
+            tmp2 = ((tmp1 == INT16_MIN  &&  tmp2 == INT16_MIN)
+                   ?
+                   INT16_MAX
+                   :
+                   (int16_t) (((int32_t) tmp1*(int32_t) tmp2 + 16384) >> 15) & 0xFFFF);
+
+            sri = gsm_sub(sri, tmp2);
+
+            tmp1 = ((tmp1 == INT16_MIN  &&  sri == INT16_MIN)
+                    ?
+                    INT16_MAX
+                    :
+                    (int16_t) (((int32_t) tmp1*(int32_t) sri + 16384) >> 15) & 0xFFFF);
+
+            v[i + 1] = gsm_add(v[i], tmp1);
+        }
+        /*endfor*/
+        *sr++ =
+        v[0] = sri;
+    }
+    /*endwhile*/
+}
+/*- End of function --------------------------------------------------------*/
+
+void gsm0610_short_term_analysis_filter(gsm0610_state_t *s,
+                                        int16_t LARc[8],
+                                        int16_t amp[GSM0610_FRAME_LEN])
+{
+    int16_t *LARpp_j;
+    int16_t *LARpp_j_1;
+    int16_t LARp[8];
+
+    LARpp_j = s->LARpp[s->j];
+    LARpp_j_1 = s->LARpp[s->j ^= 1];
+
+    decode_log_area_ratios(LARc, LARpp_j);
+
+    coefficients_0_12(LARpp_j_1, LARpp_j, LARp);
+    larp_to_rp(LARp);
+    short_term_analysis_filtering(s, LARp, 13, amp);
+
+    coefficients_13_26(LARpp_j_1, LARpp_j, LARp);
+    larp_to_rp(LARp);
+    short_term_analysis_filtering(s, LARp, 14, amp + 13);
+
+    coefficients_27_39(LARpp_j_1, LARpp_j, LARp);
+    larp_to_rp(LARp);
+    short_term_analysis_filtering(s, LARp, 13, amp + 27);
+
+    coefficients_40_159(LARpp_j, LARp);
+    larp_to_rp(LARp);
+    short_term_analysis_filtering(s, LARp, 120, amp + 40);
+}
+/*- End of function --------------------------------------------------------*/
+
+void gsm0610_short_term_synthesis_filter(gsm0610_state_t *s,
+                                         int16_t LARcr[8],
+                                         int16_t wt[GSM0610_FRAME_LEN],
+                                         int16_t amp[GSM0610_FRAME_LEN])
+{
+    int16_t *LARpp_j;
+    int16_t *LARpp_j_1;
+    int16_t LARp[8];
+
+    LARpp_j = s->LARpp[s->j];
+    LARpp_j_1 = s->LARpp[s->j ^= 1];
+
+    decode_log_area_ratios(LARcr, LARpp_j);
+
+    coefficients_0_12(LARpp_j_1, LARpp_j, LARp);
+    larp_to_rp(LARp);
+    short_term_synthesis_filtering(s, LARp, 13, wt, amp);
+
+    coefficients_13_26(LARpp_j_1, LARpp_j, LARp);
+    larp_to_rp(LARp);
+    short_term_synthesis_filtering(s, LARp, 14, wt + 13, amp + 13);
+
+    coefficients_27_39(LARpp_j_1, LARpp_j, LARp);
+    larp_to_rp(LARp);
+    short_term_synthesis_filtering(s, LARp, 13, wt + 27, amp + 27);
+
+    coefficients_40_159(LARpp_j, LARp);
+    larp_to_rp(LARp);
+    short_term_synthesis_filtering(s, LARp, 120, wt + 40, amp + 40);
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/ima_adpcm.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/ima_adpcm.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,387 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * ima_adpcm.c - Conversion routines between linear 16 bit PCM data and
+ *                 IMA/DVI/Intel ADPCM format.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2001, 2004 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: ima_adpcm.c,v 1.18 2006/11/30 15:41:47 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <inttypes.h>
+#include <string.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/ima_adpcm.h"
+
+/*
+ * Intel/DVI ADPCM coder/decoder.
+ *
+ * The algorithm for this coder was taken from the IMA Compatability Project
+ * proceedings, Vol 2, Number 2; May 1992.
+ *
+ * The RTP payload specs. reference a variant of DVI, called VDVI. This attempts to
+ * further compresses, in a variable bit rate manner, by expressing the 4 bit codes
+ * from the DVI codec as:
+ *
+ *  0 00
+ *  1 010
+ *  2 1100
+ *  3 11100
+ *  4 111100
+ *  5 1111100
+ *  6 11111100
+ *  7 11111110
+ *  8 10
+ *  9 011
+ * 10 1101
+ * 11 11101
+ * 12 111101
+ * 13 1111101
+ * 14 11111101
+ * 15 11111111
+ *
+ * Any left over bits in the last octet of an encoded burst are set to one.
+ */
+
+/* Intel ADPCM step variation table */
+static const int step_size[89] =
+{
+        7,     8,     9,   10,     11,    12,    13,    14,    16,    17,
+       19,    21,    23,   25,     28,    31,    34,    37,    41,    45,
+       50,    55,    60,   66,     73,    80,    88,    97,   107,   118,
+      130,   143,   157,   173,   190,   209,   230,   253,   279,   307,
+      337,   371,   408,   449,   494,   544,   598,   658,   724,   796,
+      876,   963,  1060,  1166,  1282,  1411,  1552,  1707,  1878,  2066,
+     2272,  2499,  2749,  3024,  3327,  3660,  4026,  4428,  4871,  5358,
+     5894,  6484,  7132,  7845,  8630,  9493, 10442, 11487, 12635, 13899,
+    15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767
+};
+
+static const int step_adjustment[8] =
+{
+    -1, -1, -1, -1, 2, 4, 6, 8
+};
+
+static const struct
+{
+    uint8_t code;
+    uint8_t bits;
+} vdvi_encode[] =
+{
+    {0x00,       2},
+    {0x02,       3},
+    {0x0C,       4},
+    {0x1C,       5},
+    {0x3C,       6},
+    {0x7C,       7},
+    {0xFC,       8},
+    {0xFE,       8},
+    {0x02,       2},
+    {0x03,       3},
+    {0x0D,       4},
+    {0x1D,       5},
+    {0x3D,       6},
+    {0x7D,       7},
+    {0xFD,       8},
+    {0xFF,       8}
+};
+
+static const struct
+{
+    uint16_t code;
+    uint16_t mask;
+    uint8_t bits;
+} vdvi_decode[] =
+{
+    {0x0000,    0xC000,     2},
+    {0x4000,    0xE000,     3},
+    {0xC000,    0xF000,     4},
+    {0xE000,    0xF800,     5},
+    {0xF000,    0xFC00,     6},
+    {0xF800,    0xFE00,     7},
+    {0xFC00,    0xFF00,     8},
+    {0xFE00,    0xFF00,     8},
+    {0x8000,    0xC000,     2},
+    {0x6000,    0xE000,     3},
+    {0xD000,    0xF000,     4},
+    {0xE800,    0xF800,     5},
+    {0xF400,    0xFC00,     6},
+    {0xFA00,    0xFE00,     7},
+    {0xFD00,    0xFF00,     8},
+    {0xFF00,    0xFF00,     8}
+};
+
+static int16_t decode(ima_adpcm_state_t *s, uint8_t adpcm)
+{
+    int e;
+    int ss;
+    int16_t linear;
+
+    /* e = (adpcm+0.5)*step/4 */
+
+    ss = step_size[s->step_index];
+    e = ss >> 3;
+    if (adpcm & 0x01)
+        e += (ss >> 2);
+    /*endif*/
+    if (adpcm & 0x02)
+        e += (ss >> 1);
+    /*endif*/
+    if (adpcm & 0x04)
+        e += ss;
+    /*endif*/
+    if (adpcm & 0x08)
+        e = -e;
+    /*endif*/
+    linear = saturate(s->last + e);
+    s->last = linear;
+    s->step_index += step_adjustment[adpcm & 0x07];
+    if (s->step_index < 0)
+        s->step_index = 0;
+    else if (s->step_index > 88)
+        s->step_index = 88;
+    /*endif*/
+    return linear;
+}
+/*- End of function --------------------------------------------------------*/
+
+static uint8_t encode(ima_adpcm_state_t *s, int16_t linear)
+{
+    int e;
+    int ss;
+    int adpcm;
+    int diff;
+    int initial_e;
+
+    ss = step_size[s->step_index];
+    initial_e =
+    e = linear - s->last;
+    diff = ss >> 3;
+    adpcm = (uint8_t) 0x00;
+    if (e < 0)
+    {
+        adpcm = (uint8_t) 0x08;
+        e = -e;
+    }
+    /*endif*/
+    if (e >= ss)
+    {
+        adpcm |= (uint8_t) 0x04;
+        e -= ss;
+    }
+    /*endif*/
+    ss >>= 1;
+    if (e >= ss)
+    {
+        adpcm |= (uint8_t) 0x02;
+        e -= ss;
+    }
+    /*endif*/
+    ss >>= 1;
+    if (e >= ss)
+    {
+        adpcm |= (uint8_t) 0x01;
+        e -= ss;
+    }
+    /*endif*/
+
+    if (initial_e < 0)
+        diff = -(diff - initial_e - e);
+    else
+        diff = diff + initial_e - e;
+    /*endif*/
+    s->last = saturate(diff + s->last);
+    s->step_index += step_adjustment[adpcm & 0x07];
+    if (s->step_index < 0)
+        s->step_index = 0;
+    else if (s->step_index > 88)
+        s->step_index = 88;
+    /*endif*/
+    return (uint8_t) adpcm;
+}
+/*- End of function --------------------------------------------------------*/
+
+ima_adpcm_state_t *ima_adpcm_init(ima_adpcm_state_t *s, int variant)
+{
+    if (s == NULL)
+    {
+        if ((s = (ima_adpcm_state_t *) malloc(sizeof(*s))) == NULL)
+            return  NULL;
+    }
+    /*endif*/
+    memset(s, 0, sizeof(*s));
+    s->variant = variant;
+    return  s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int ima_adpcm_release(ima_adpcm_state_t *s)
+{
+    free(s);
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+int ima_adpcm_decode(ima_adpcm_state_t *s,
+                     int16_t amp[],
+                     const uint8_t ima_data[],
+                     int ima_bytes)
+{
+    int i;
+    int j;
+    int samples;
+    uint16_t code;
+
+    samples = 0;
+    if (s->variant == IMA_ADPCM_VDVI)
+    {
+        code = 0;
+        s->bits = 0;
+        for (i = 0;  ;  )
+        {
+            if (s->bits <= 8)
+            {
+                if (i >= ima_bytes)
+                    break;
+                /*endif*/
+                code |= ((uint16_t) ima_data[i++] << (8 - s->bits));
+                s->bits += 8;
+            }
+            /*endif*/
+            for (j = 0;  j < 8;  j++)
+            {
+                if ((vdvi_decode[j].mask & code) == vdvi_decode[j].code)
+                    break;
+                if ((vdvi_decode[j + 8].mask & code) == vdvi_decode[j + 8].code)
+                {
+                    j += 8;
+                    break;
+                }
+                /*endif*/
+            }
+            /*endfor*/
+            amp[samples++] = decode(s, (uint8_t) j);
+            code <<= vdvi_decode[j].bits;
+            s->bits -= vdvi_decode[j].bits;
+        }
+        /*endfor*/
+        /* Use up the remanents of the last octet */
+        while (s->bits > 0)
+        {
+            for (j = 0;  j < 8;  j++)
+            {
+                if ((vdvi_decode[j].mask & code) == vdvi_decode[j].code)
+                    break;
+                /*endif*/
+                if ((vdvi_decode[j + 8].mask & code) == vdvi_decode[j + 8].code)
+                {
+                    j += 8;
+                    break;
+                }
+                /*endif*/
+            }
+            /*endfor*/
+            if (vdvi_decode[j].bits > s->bits)
+                break;
+            /*endif*/
+            amp[samples++] = decode(s, (uint8_t) j);
+            code <<= vdvi_decode[j].bits;
+            s->bits -= vdvi_decode[j].bits;
+        }
+        /*endfor*/
+    }
+    else
+    {
+        for (i = 0;  i < ima_bytes;  i++)
+        {
+            amp[samples++] = decode(s, ima_data[i] & 0xF);
+            amp[samples++] = decode(s, (ima_data[i] >> 4) & 0xF);
+        }
+        /*endwhile*/
+    }
+    /*endif*/
+    return samples;
+}
+/*- End of function --------------------------------------------------------*/
+
+int ima_adpcm_encode(ima_adpcm_state_t *s,
+                     uint8_t ima_data[],
+                     const int16_t amp[],
+                     int len)
+{
+    int i;
+    int bytes;
+    uint8_t code;
+
+    bytes = 0;
+    if (s->variant == IMA_ADPCM_VDVI)
+    {
+        s->bits = 0;
+        for (i = 0;  i < len;  i++)
+        {
+            code = encode(s, amp[i]);
+            s->ima_byte = (s->ima_byte << vdvi_encode[code].bits) | vdvi_encode[code].code;
+            s->bits += vdvi_encode[code].bits;
+            if (s->bits >= 8)
+            {
+                s->bits -= 8;
+                ima_data[bytes++] = (uint8_t) (s->ima_byte >> s->bits);
+            }
+            /*endif*/
+        }
+        /*endfor*/
+        if (s->bits)
+        {
+            ima_data[bytes++] = (uint8_t) (((s->ima_byte << 8) | 0xFF) >> s->bits);
+        }
+        /*endif*/
+    }
+    else
+    {
+        for (i = 0;  i < len;  i++)
+        {
+            s->ima_byte = (uint8_t) ((s->ima_byte >> 4) | (encode(s, amp[i]) << 4));
+            if ((s->bits++ & 1))
+                ima_data[bytes++] = (uint8_t) s->ima_byte;
+            /*endif*/
+        }
+        /*endfor*/
+    }
+    /*endif*/
+    return  bytes;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/libvoipcodecs.dsp
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/libvoipcodecs.dsp	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,241 @@
+# Microsoft Developer Studio Project File - Name="voipcodecs" - Package Owner=<4>
+# Microsoft Developer Studio Generated Build File, Format Version 6.00
+# ** DO NOT EDIT **
+
+# TARGTYPE "Win32 (x86) Dynamic-Link Library" 0x0102
+
+CFG=voipcodecs - Win32 Debug
+!MESSAGE This is not a valid makefile. To build this project using NMAKE,
+!MESSAGE use the Export Makefile command and run
+!MESSAGE 
+!MESSAGE NMAKE /f "voipcodecs.mak".
+!MESSAGE 
+!MESSAGE You can specify a configuration when running NMAKE
+!MESSAGE by defining the macro CFG on the command line. For example:
+!MESSAGE 
+!MESSAGE NMAKE /f "voipcodecs.mak" CFG="voipcodecs - Win32 Debug"
+!MESSAGE 
+!MESSAGE Possible choices for configuration are:
+!MESSAGE 
+!MESSAGE "voipcodecs - Win32 Release" (based on "Win32 (x86) Dynamic-Link Library")
+!MESSAGE "voipcodecs - Win32 Debug" (based on "Win32 (x86) Dynamic-Link Library")
+!MESSAGE 
+
+# Begin Project
+# PROP AllowPerConfigDependencies 0
+# PROP Scc_ProjName ""
+# PROP Scc_LocalPath ""
+CPP=cl.exe
+MTL=midl.exe
+RSC=rc.exe
+
+!IF  "$(CFG)" == "voipcodecs - Win32 Release"
+
+# PROP BASE Use_MFC 0
+# PROP BASE Use_Debug_Libraries 0
+# PROP BASE Output_Dir "Release"
+# PROP BASE Intermediate_Dir "Release"
+# PROP BASE Target_Dir ""
+# PROP Use_MFC 0
+# PROP Use_Debug_Libraries 0
+# PROP Output_Dir "Release"
+# PROP Intermediate_Dir "Release"
+# PROP Ignore_Export_Lib 0
+# PROP Target_Dir ""
+# ADD BASE CPP /nologo /MT /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D HAVE_TGMATH_H /YX /FD /c
+# ADD CPP /nologo /MT /W3 /GX /Zi /O2 /I "." /I "..\include" /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D HAVE_TGMATH_H /D "_WINDLL" /FR /FD /c
+# SUBTRACT CPP /YX
+# ADD BASE MTL /nologo /D "NDEBUG" /mktyplib203 /win32
+# ADD MTL /nologo /D "NDEBUG" /mktyplib203 /win32
+# ADD BASE RSC /l 0x409 /d "NDEBUG"
+# ADD RSC /l 0x409 /d "NDEBUG"
+BSC32=bscmake.exe
+# ADD BASE BSC32 /nologo
+# ADD BSC32 /nologo
+LINK32=link.exe
+# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /dll /machine:I386
+# ADD LINK32 kernel32.lib ws2_32.lib winmm.lib /nologo /dll /map /debug /machine:I386 /out:"Release/libvoipcodecs.dll"
+
+!ELSEIF  "$(CFG)" == "voipcodecs - Win32 Debug"
+
+# PROP BASE Use_MFC 0
+# PROP BASE Use_Debug_Libraries 1
+# PROP BASE Output_Dir "Debug"
+# PROP BASE Intermediate_Dir "Debug"
+# PROP BASE Target_Dir ""
+# PROP Use_MFC 0
+# PROP Use_Debug_Libraries 1
+# PROP Output_Dir "Debug"
+# PROP Intermediate_Dir "Debug"
+# PROP Ignore_Export_Lib 0
+# PROP Target_Dir ""
+# ADD BASE CPP /nologo /MTd /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D HAVE_TGMATH_H /YX /FD /GZ /c
+# ADD CPP /nologo /MTd /W3 /Gm /GX /Zi /Od /I "." /I "..\include" /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D HAVE_TGMATH_H /FR /FD /GZ /c
+# SUBTRACT CPP /WX /YX
+# ADD BASE MTL /nologo /D "_DEBUG" /mktyplib203 /win32
+# ADD MTL /nologo /D "_DEBUG" /mktyplib203 /win32
+# ADD BASE RSC /l 0x409 /d "_DEBUG"
+# ADD RSC /l 0x409 /d "_DEBUG"
+BSC32=bscmake.exe
+# ADD BASE BSC32 /nologo
+# ADD BSC32 /nologo
+LINK32=link.exe
+# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /dll /debug /machine:I386 /pdbtype:sept
+# ADD LINK32 kernel32.lib ws2_32.lib winmm.lib /nologo /dll /incremental:no /map /debug /machine:I386 /out:"Debug/libvoipcodecs.dll" /pdbtype:sept
+# SUBTRACT LINK32 /nodefaultlib
+
+!ENDIF 
+
+# Begin Target
+
+# Name "voipcodecs - Win32 Release"
+# Name "voipcodecs - Win32 Debug"
+# Begin Group "Source Files"
+# Begin Source File
+
+SOURCE=.\bit_operations.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\bitstream.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\g711.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\g722_encode.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\g722_decode.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\g726.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\gsm0610_decode.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\gsm0610_encode.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\gsm0610_long_term.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\gsm0610_lpc.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\gsm0610_preprocess.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\gsm0610_rpe.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\gsm0610_short_term.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\ima_adpcm.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\lpc10_analyse.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\lpc10_decode.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\lpc10_encode.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\lpc10_placev.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\lpc10_voicing.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\oki_adpcm.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\vector_int.c
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/bit_operations.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/bitstream.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/dc_restore.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/g711.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/g722.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/g726.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/gsm0610.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/ima_adpcm.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/lpc10.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/oki_adpcm.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/telephony.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs/vector_int.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs.h
+# End Source File
+# Begin Source File
+
+SOURCE=.\voipcodecs.h
+# End Source File
+# End Group
+
+# Begin Group "Resource Files"
+
+# PROP Default_Filter "ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe"
+# End Group
+# End Target
+# End Project

Added: freeswitch/trunk/libs/voipcodecs/src/libvoipcodecs.sln
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/libvoipcodecs.sln	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,29 @@
+
+Microsoft Visual Studio Solution File, Format Version 9.00
+# Visual Studio 2005
+Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "libvoipcodecs", "libvoipcodecs.vcproj", "{CF70F278-3364-4395-A2E1-23501C9B8AD2}"
+	ProjectSection(ProjectDependencies) = postProject
+		{1CED5987-A529-46DC-B30F-870D85FF9C94} = {1CED5987-A529-46DC-B30F-870D85FF9C94}
+	EndProjectSection
+EndProject
+Project("{8BC9CEB8-8B4A-11D0-8D11-00A0C91BC942}") = "at_dictionary_gen", "src\msvc\at_dictionary_gen.vcproj", "{1CED5987-A529-46DC-B30F-870D85FF9C94}"
+EndProject
+Global
+	GlobalSection(SolutionConfigurationPlatforms) = preSolution
+		Debug|Win32 = Debug|Win32
+		Release|Win32 = Release|Win32
+	EndGlobalSection
+	GlobalSection(ProjectConfigurationPlatforms) = postSolution
+		{CF70F278-3364-4395-A2E1-23501C9B8AD2}.Debug|Win32.ActiveCfg = Debug|Win32
+		{CF70F278-3364-4395-A2E1-23501C9B8AD2}.Debug|Win32.Build.0 = Debug|Win32
+		{CF70F278-3364-4395-A2E1-23501C9B8AD2}.Release|Win32.ActiveCfg = Release|Win32
+		{CF70F278-3364-4395-A2E1-23501C9B8AD2}.Release|Win32.Build.0 = Release|Win32
+		{1CED5987-A529-46DC-B30F-870D85FF9C94}.Debug|Win32.ActiveCfg = Debug|Win32
+		{1CED5987-A529-46DC-B30F-870D85FF9C94}.Debug|Win32.Build.0 = Debug|Win32
+		{1CED5987-A529-46DC-B30F-870D85FF9C94}.Release|Win32.ActiveCfg = Release|Win32
+		{1CED5987-A529-46DC-B30F-870D85FF9C94}.Release|Win32.Build.0 = Release|Win32
+	EndGlobalSection
+	GlobalSection(SolutionProperties) = preSolution
+		HideSolutionNode = FALSE
+	EndGlobalSection
+EndGlobal

Added: freeswitch/trunk/libs/voipcodecs/src/lpc10_analyse.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/lpc10_analyse.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,712 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * lpc10_analyse.c - LPC10 low bit rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the U.S. Department of Defense reference
+ * implementation of the LPC-10 2400 bps Voice Coder. They do not
+ * exert copyright claims on their code, and it may be freely used.
+ *
+ * $Id: lpc10_analyse.c,v 1.13 2007/01/03 14:15:35 steveu Exp $
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <inttypes.h>
+#include <memory.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/lpc10.h"
+
+#include "lpc10_encdecs.h"
+
+static __inline__ float energyf(float amp[], int len)
+{
+    int i;
+    float rms;
+
+    rms = 0.0f;
+    for (i = 0;  i < len;  i++)
+        rms += amp[i]*amp[i];
+    rms = sqrtf(rms/len);
+    return rms;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void remove_dc_bias(float speech[], int len, float sigout[])
+{
+    float bias;
+    int i;
+
+    bias = 0.0f;
+    for (i = 0;  i < len;  i++)
+        bias += speech[i];
+    bias /= len;
+    for (i = 0; i < len;  i++)
+        sigout[i] = speech[i] - bias;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void eval_amdf(float speech[],
+                       int32_t lpita,
+                       const int32_t tau[], 
+                       int32_t ltau,
+                       int32_t maxlag,
+                       float amdf[],
+                       int32_t *minptr,
+                       int32_t *maxptr)
+{
+    float sum;
+    int i;
+    int j;
+    int n1;
+    int n2;
+
+    *minptr = 0;
+    *maxptr = 0;
+    for (i = 0;  i < ltau;  i++)
+    {
+        n1 = (maxlag - tau[i])/2 + 1;
+        n2 = n1 + lpita - 1;
+        sum = 0.0f;
+        for (j = n1;  j <= n2;  j += 4)
+            sum += fabsf(speech[j - 1] - speech[j + tau[i] - 1]);
+        amdf[i] = sum;
+        if (amdf[i] < amdf[*minptr])
+            *minptr = i;
+        if (amdf[i] > amdf[*maxptr])
+            *maxptr = i;
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+static void eval_highres_amdf(float speech[],
+                              int32_t lpita,
+                              const int32_t tau[], 
+                              int32_t ltau,
+                              float amdf[],
+                              int32_t *minptr,
+                              int32_t *maxptr,
+                              int32_t *mintau)
+{
+    float amdf2[6];
+    int32_t tau2[6];
+    int32_t minp2;
+    int32_t ltau2;
+    int32_t maxp2;
+    int32_t minamd;
+    int i;
+    int i2;
+    int ptr;
+
+    /* Compute full AMDF using log spaced lags, find coarse minimum */
+    eval_amdf(speech, lpita, tau, ltau, tau[ltau - 1], amdf, minptr, maxptr);
+    *mintau = tau[*minptr];
+    minamd = (int32_t) amdf[*minptr];
+
+    /* Build table containing all lags within +/- 3 of the AMDF minimum,
+       excluding all that have already been computed */
+    ltau2 = 0;
+    ptr = *minptr - 2;
+    i2 = min(*mintau + 4, tau[ltau - 1]);
+    for (i = max(*mintau - 3, 41);  i < i2;  i++)
+    {
+        while (tau[ptr] < i)
+            ptr++;
+        if (tau[ptr] != i)
+            tau2[ltau2++] = i;
+    }
+    /* Compute AMDF of the new lags, if there are any, and choose one
+       if it is better than the coarse minimum */
+    if (ltau2 > 0)
+    {
+        eval_amdf(speech, lpita, tau2, ltau2, tau[ltau - 1], amdf2, &minp2, &maxp2);
+        if (amdf2[minp2] < (float) minamd)
+        {
+            *mintau = tau2[minp2];
+            minamd = (int32_t) amdf2[minp2];
+        }
+    }
+    /* Check one octave up, if there are any lags not yet computed */
+    if (*mintau >= 80)
+    {
+        i = *mintau/2;
+        if ((i & 1) == 0)
+        {
+            ltau2 = 2;
+            tau2[0] = i - 1;
+            tau2[1] = i + 1;
+        }
+        else
+        {
+            ltau2 = 1;
+            tau2[0] = i;
+        }
+        eval_amdf(speech, lpita, tau2, ltau2, tau[ltau - 1], amdf2, &minp2, &maxp2);
+        if (amdf2[minp2] < (float) minamd)
+        {
+            *mintau = tau2[minp2];
+            minamd = (int32_t) amdf2[minp2];
+            *minptr -= 20;
+        }
+    }
+    /* Force minimum of the AMDF array to the high resolution minimum */
+    amdf[*minptr] = (float) minamd;
+    /* Find maximum of AMDF within 1/2 octave of minimum */
+    *maxptr = max(*minptr - 5, 0);
+    i2 = min(*minptr + 6, ltau);
+    for (i = *maxptr;  i < i2;  i++)
+    {
+        if (amdf[i] > amdf[*maxptr])
+            *maxptr = i;
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+static void dynamic_pitch_tracking(lpc10_encode_state_t *s,
+                                   float amdf[],
+                                   int32_t ltau,
+                                   int32_t *minptr,
+                                   int32_t voice,
+                                   int32_t *pitch,
+                                   int32_t *midx)
+{
+    int32_t pbar;
+    float sbar;
+    int32_t path[2];
+    int32_t i;
+    int32_t j;
+    float alpha;
+    float minsc;
+    float maxsc;
+
+    /* Calculate the confidence factor ALPHA, used as a threshold slope in */
+    /* SEESAW.  If unvoiced, set high slope so that every point in P array */
+    /*is marked as a potential pitch frequency.  A scaled up version (ALPHAX )*/
+    /* is used to maintain arithmetic precision. */
+    if (voice == 1)
+        s->alphax = s->alphax*0.75f + amdf[*minptr - 1]*0.5f;
+    else
+        s->alphax *= 0.984375f;
+    alpha = s->alphax/16;
+    if (voice == 0  &&  s->alphax < 128.0f)
+        alpha = 8.0f;
+    /* SEESAW: Construct a pitch pointer array and intermediate winner function */
+    /* Left to right pass: */
+    s->p[s->ipoint][0] = 1;
+    pbar = 1;
+    sbar = s->s[0];
+    for (i = 0;  i < ltau;  i++)
+    {
+        sbar += alpha;
+        if (sbar < s->s[i])
+        {
+            s->s[i] = sbar;
+        }
+        else
+        {
+            pbar = i + 1;
+            sbar = s->s[i];
+        }
+        s->p[s->ipoint][i] = pbar;
+    }
+    /* Right to left pass: */
+    sbar = s->s[pbar - 1];
+    for (i = pbar - 2;  i >= 0;  i--)
+    {
+        sbar += alpha;
+        if (sbar < s->s[i])
+        {
+            s->s[i] = sbar;
+            s->p[s->ipoint][i] = pbar;
+        }
+        else
+        {
+            pbar = s->p[s->ipoint][i];
+            i = pbar - 1;
+            sbar = s->s[i];
+        }
+    }
+    /* Update S using AMDF */
+    /* Find maximum, minimum, and location of minimum */
+    s->s[0] += amdf[0]/2;
+    minsc = s->s[0];
+    maxsc = minsc;
+    *midx = 1;
+    for (i = 1;  i < ltau;  i++)
+    {
+        s->s[i] += amdf[i]/2;
+        if (s->s[i] > maxsc)
+            maxsc = s->s[i];
+        if (s->s[i] < minsc)
+        {
+            *midx = i + 1;
+            minsc = s->s[i];
+        }
+    }
+    /* Subtract MINSC from S to prevent overflow */
+    for (i = 0;  i < ltau;  i++)
+        s->s[i] -= minsc;
+    maxsc -= minsc;
+    /* Use higher octave pitch if significant null there */
+    j = 0;
+    for (i = 20;  i <= 40;  i += 10)
+    {
+        if (*midx > i)
+        {
+            if (s->s[*midx - i - 1] < maxsc / 4)
+                j = i;
+        }
+    }
+    *midx -= j;
+    /* TRACE: look back two frames to find minimum cost pitch estimate */
+    *pitch = *midx;
+    for (i = 0, j = s->ipoint;  i < 2;  i++, j++)
+    {
+        *pitch = s->p[j & 1][*pitch - 1];
+        path[i] = *pitch;
+    }
+
+    /* The following statement subtracts one from IPOINT, mod DEPTH.  I */
+    /* think the author chose to add DEPTH-1, instead of subtracting 1, */
+    /* because then it will work even if MOD doesn't work as desired on */
+    /* negative arguments. */
+    s->ipoint = (s->ipoint + 1) & 1;
+}
+/*- End of function --------------------------------------------------------*/
+
+/* Detection of onsets in (or slightly preceding) the futuremost frame of speech. */
+static void onset(lpc10_encode_state_t *s,
+                   float *pebuf,
+                   int32_t osbuf[],
+                   int32_t *osptr,
+                   int32_t oslen,
+                   int32_t sbufl,
+                   int32_t sbufh,
+                   int32_t lframe)
+{
+    int32_t i;
+    float r1;
+    float l2sum2;
+
+    pebuf -= sbufl;
+
+    if (s->hyst)
+        s->lasti -= lframe;
+    for (i = sbufh - lframe + 1;  i <= sbufh;  i++)
+    {
+        /* Compute FPC; Use old FPC on divide by zero; Clamp FPC to +/- 1. */
+        s->n = (pebuf[i]*pebuf[i - 1] + s->n*63.0f)/64.0f;
+        /* Computing 2nd power */
+        r1 = pebuf[i - 1];
+        s->d__ = (r1*r1 + s->d__*63.0f)/64.0f;
+        if (s->d__ != 0.0f)
+        {
+            if (fabsf(s->n) > s->d__)
+                s->fpc = r_sign(1.0f, s->n);
+            else
+                s->fpc = s->n/s->d__;
+        }
+        /* Filter FPC */
+        l2sum2 = s->l2buf[s->l2ptr1 - 1];
+        s->l2sum1 = s->l2sum1 - s->l2buf[s->l2ptr2 - 1] + s->fpc;
+        s->l2buf[s->l2ptr2 - 1] = s->l2sum1;
+        s->l2buf[s->l2ptr1 - 1] = s->fpc;
+        s->l2ptr1 = (s->l2ptr1 & 0xF) + 1;
+        s->l2ptr2 = (s->l2ptr2 & 0xF) + 1;
+        if (fabsf(s->l2sum1 - l2sum2) > 1.7f)
+        {
+            if (!s->hyst)
+            {
+                /* Ignore if buffer full */
+                if (*osptr <= oslen)
+                {
+                    osbuf[*osptr - 1] = i - 9;
+                    (*osptr)++;
+                }
+                s->hyst = TRUE;
+            }
+            s->lasti = i;
+            /* After one onset detection, at least OSHYST sample times must go */
+            /* by before another is allowed to occur. */
+        }
+        else if (s->hyst  &&  i - s->lasti >= 10)
+        {
+            s->hyst = FALSE;
+        }
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+/* Load a covariance matrix. */
+static void mload(int32_t order, int32_t awins, int32_t awinf, float speech[], float phi[], float psi[])
+{
+    int32_t start;
+    int i;
+    int r;
+
+    start = awins + order;
+    for (r = 1;  r <= order;  r++)
+    {
+        phi[r - 1] = 0.0f;
+        for (i = start;  i <= awinf;  i++)
+            phi[r - 1] += speech[i - 2]*speech[i - r - 1];
+    }
+
+    /* Load last element of vector PSI */
+    psi[order - 1] = 0.0f;
+    for (i = start - 1;  i < awinf;  i++)
+        psi[order - 1] += speech[i]*speech[i - order];
+    /* End correct to get additional columns of phi */
+    for (r = 1;  r < order;  r++)
+    {
+        for (i = 1;  i <= r;  i++)
+        {
+            phi[i*order + r] = phi[(i - 1)*order + r - 1]
+                             - speech[awinf - (r + 1)]*speech[awinf - (i + 1)]
+                             + speech[start - (r + 2)]*speech[start - (i + 2)];
+        }
+    }
+    /* End correct to get additional elements of PSI */
+    for (i = 0;  i < order - 1;  i++)
+    {
+        psi[i] = phi[i + 1]
+               - speech[start - 2]*speech[start - i - 3]
+               + speech[awinf - 1]*speech[awinf - i - 2];
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+/* Preemphasize speech with a single-zero filter. */
+/* (When coef = .9375, preemphasis is as in LPC43.) */
+static float preemp(float inbuf[], float pebuf[], int nsamp, float coeff, float z)
+{
+    float temp;
+    int i;
+
+    for (i = 0;  i < nsamp;  i++)
+    {
+        temp = inbuf[i] - coeff*z;
+        z = inbuf[i];
+        pebuf[i] = temp;
+    }
+    return z;
+}
+/*- End of function --------------------------------------------------------*/
+
+/* Invert a covariance matrix using Choleski decomposition method. */
+static void invert(int32_t order, float phi[], float psi[], float rc[])
+{
+    float r1;
+    int32_t i;
+    int32_t j;
+    int32_t k;
+    float v[10][10];
+
+    for (j = 0;  j < order;  j++)
+    {
+        for (i = j;  i < order;  i++)
+            v[j][i] = phi[i + j*order];
+        for (k = 0;  k < j;  k++)
+        {
+            r1 = v[k][j]*v[k][k];
+            for (i = j;  i <= order;  i++)
+                v[j][i] -= v[k][i]*r1;
+        }
+        /* Compute intermediate results, which are similar to RC's */
+        if (fabsf(v[j][j]) < 1.0e-10f)
+        {
+            for (i = j;  i < order;  i++)
+                rc[i] = 0.0f;
+            return;
+        }
+        rc[j] = psi[j];
+        for (k = 0;  k < j;  k++)
+            rc[j] -= rc[k]*v[k][j];
+        v[j][j] = 1.0f/v[j][j];
+        rc[j] *= v[j][j];
+        r1 = min(rc[j], 0.999f);
+        rc[j] = max(r1, -0.999f);
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+/* Check RC's, repeat previous frame's RC's if unstable */
+static int rcchk(int order, float rc1f[], float rc2f[])
+{
+    int i;
+
+    for (i = 0;  i < order;  i++)
+    {
+        if (fabsf(rc2f[i]) > 0.99f)
+        {
+            for (i = 0;  i < order;  i++)
+                rc2f[i] = rc1f[i];
+            break;
+        }
+    }
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void lpfilt(float inbuf[], float lpbuf[], int32_t len, int32_t nsamp)
+{
+    int32_t j;
+    float t;
+
+    /* 31 point equiripple FIR LPF */
+    /* Linear phase, delay = 15 samples */
+    /* Passband:  ripple = 0.25 dB, cutoff =  800 Hz */
+    /* Stopband:  atten. =  40. dB, cutoff = 1240 Hz */
+
+    for (j = len - nsamp;  j < len;  j++)
+    {
+        t = (inbuf[j] + inbuf[j - 30]) * -0.0097201988f;
+        t += (inbuf[j - 1] + inbuf[j - 29]) * -0.0105179986f;
+        t += (inbuf[j - 2] + inbuf[j - 28]) * -0.0083479648f;
+        t += (inbuf[j - 3] + inbuf[j - 27]) * 5.860774e-4f;
+        t += (inbuf[j - 4] + inbuf[j - 26]) * 0.0130892089f;
+        t += (inbuf[j - 5] + inbuf[j - 25]) * 0.0217052232f;
+        t += (inbuf[j - 6] + inbuf[j - 24]) * 0.0184161253f;
+        t += (inbuf[j - 7] + inbuf[j - 23]) * 3.39723e-4f;
+        t += (inbuf[j - 8] + inbuf[j - 22]) * -0.0260797087f;
+        t += (inbuf[j - 9] + inbuf[j - 21]) * -0.0455563702f;
+        t += (inbuf[j - 10] + inbuf[j - 20]) * -0.040306855f;
+        t += (inbuf[j - 11] + inbuf[j - 19]) * 5.029835e-4f;
+        t += (inbuf[j - 12] + inbuf[j - 18]) * 0.0729262903f;
+        t += (inbuf[j - 13] + inbuf[j - 17]) * 0.1572008878f;
+        t += (inbuf[j - 14] + inbuf[j - 16]) * 0.2247288674f;
+        t += inbuf[j - 15] * 0.250535965f;
+        lpbuf[j] = t;
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+/* 2nd order inverse filter, speech is decimated 4:1 */
+static void ivfilt(float lpbuf[], float ivbuf[], int32_t len, int32_t nsamp, float ivrc[])
+{
+    int32_t i;
+    int32_t j;
+    int32_t k;
+    float r[3];
+    float pc1;
+    float pc2;
+
+    /* Calculate autocorrelations */
+    for (i = 1;  i <= 3;  i++)
+    {
+        r[i - 1] = 0.0f;
+        k = (i - 1) << 2;
+        for (j = (i << 2) + len - nsamp;  j <= len;  j += 2)
+            r[i - 1] += lpbuf[j - 1]*lpbuf[j - k - 1];
+    }
+    /* Calculate predictor coefficients */
+    pc1 = 0.0f;
+    pc2 = 0.0f;
+    ivrc[0] = 0.0f;
+    ivrc[1] = 0.0f;
+    if (r[0] > 1.0e-10f)
+    {
+        ivrc[0] = r[1]/r[0];
+        ivrc[1] = (r[2] - ivrc[0]*r[1])/(r[0] - ivrc[0]*r[1]);
+        pc1 = ivrc[0] - ivrc[0]*ivrc[1];
+        pc2 = ivrc[1];
+    }
+    /* Inverse filter LPBUF into IVBUF */
+    for (i = len - nsamp;  i < len;  i++)
+        ivbuf[i] = lpbuf[i] - pc1*lpbuf[i - 4] - pc2*lpbuf[i - 8];
+}
+/*- End of function --------------------------------------------------------*/
+
+void lpc10_analyse(lpc10_encode_state_t *s, float speech[], int32_t voice[], int32_t *pitch, float *rms, float rc[])
+{
+    static const int32_t tau[60] =
+    {
+        20, 21, 22, 23, 24, 25, 26, 27, 28, 29, 30, 31, 32, 33, 34, 
+        35, 36, 37, 38, 39, 40, 42, 44, 46, 48, 50, 52, 54, 56, 58,
+        60, 62, 64, 66, 68, 70, 72, 74, 76, 78, 80, 84, 88, 92, 96,
+        100, 104, 108, 112, 116, 120, 124, 128, 132, 136, 140, 144,
+        148, 152, 156
+    };
+    static const int32_t buflim[4] =
+    {
+        181, 720, 25, 720
+    };
+    static const float precoef = 0.9375f;
+
+    float amdf[60];
+    float abuf[156];
+    float ivrc[2];
+    float temp;
+    float phi[100]    /* was [10][10] */;
+    float psi[10];
+    int32_t half;
+    int32_t midx;
+    int32_t ewin[3][2];
+    int32_t i;
+    int32_t j;
+    int32_t lanal;
+    int32_t ipitch;
+    int32_t mintau;
+    int32_t minptr;
+    int32_t maxptr;
+
+    /* Calculations are done on future frame due to requirements
+       of the pitch tracker.  Delay RMS and RC's 2 frames to give
+       current frame parameters on return. */
+
+    for (i = 0;  i <= 720 - LPC10_SAMPLES_PER_FRAME - 181;  i++)
+    {
+        s->inbuf[i] = s->inbuf[LPC10_SAMPLES_PER_FRAME + i];
+        s->pebuf[i] = s->pebuf[LPC10_SAMPLES_PER_FRAME + i];
+    }
+    for (i = 0;  i <= 540 - LPC10_SAMPLES_PER_FRAME - 229;  i++)
+        s->ivbuf[i] = s->ivbuf[LPC10_SAMPLES_PER_FRAME + i];
+    for (i = 0;  i <= 720 - LPC10_SAMPLES_PER_FRAME - 25;  i++)
+        s->lpbuf[i] = s->lpbuf[LPC10_SAMPLES_PER_FRAME + i];
+    for (i = 0, j = 0;  i < s->osptr - 1;  i++)
+    {
+        if (s->osbuf[i] > LPC10_SAMPLES_PER_FRAME)
+            s->osbuf[j++] = s->osbuf[i] - LPC10_SAMPLES_PER_FRAME;
+    }
+    s->osptr = j + 1;
+    s->voibuf[0][0] = s->voibuf[1][0];
+    s->voibuf[0][1] = s->voibuf[1][1];
+    for (i = 0;  i < 2;  i++)
+    {
+        s->vwin[i][0] = s->vwin[i + 1][0] - LPC10_SAMPLES_PER_FRAME;
+        s->vwin[i][1] = s->vwin[i + 1][1] - LPC10_SAMPLES_PER_FRAME;
+        s->awin[i][0] = s->awin[i + 1][0] - LPC10_SAMPLES_PER_FRAME;
+        s->awin[i][1] = s->awin[i + 1][1] - LPC10_SAMPLES_PER_FRAME;
+        s->obound[i] = s->obound[i + 1];
+        s->voibuf[i + 1][0] = s->voibuf[i + 2][0];
+        s->voibuf[i + 1][1] = s->voibuf[i + 2][1];
+        s->rmsbuf[i] = s->rmsbuf[i + 1];
+        for (j = 0;  j < LPC10_ORDER;  j++)
+            s->rcbuf[i][j] = s->rcbuf[i + 1][j];
+    }
+    /* If the average value in the frame was over 1/4096 (after current
+       BIAS correction), then subtract that much more from samples in the
+       next frame.  If the average value in the frame was under
+       -1/4096, add 1/4096 more to samples in next frame.  In all other
+       cases, keep BIAS the same. */
+    temp = 0.0f;
+    for (i = 0;  i < LPC10_SAMPLES_PER_FRAME;  i++)
+    {
+        s->inbuf[720 - 2*LPC10_SAMPLES_PER_FRAME + i] = speech[i]*4096.0f - s->bias;
+        temp += s->inbuf[720 - 2*LPC10_SAMPLES_PER_FRAME + i];
+    }
+    if (temp > (float) LPC10_SAMPLES_PER_FRAME)
+        s->bias++;
+    else if (temp < (float) (-LPC10_SAMPLES_PER_FRAME))
+        s->bias--;
+    /* Place voicing window */
+    i = 721 - LPC10_SAMPLES_PER_FRAME;
+    s->zpre = preemp(&s->inbuf[i - 181], &s->pebuf[i - 181], LPC10_SAMPLES_PER_FRAME, precoef, s->zpre);
+    onset(s, s->pebuf, s->osbuf, &s->osptr, 10, 181, 720, LPC10_SAMPLES_PER_FRAME);
+
+    lpc10_placev(s->osbuf, &s->osptr, 10, &s->obound[2], s->vwin, 3, LPC10_SAMPLES_PER_FRAME, 90, 156, 307, 462);
+    /* The Pitch Extraction algorithm estimates the pitch for a frame
+       of speech by locating the minimum of the average magnitude difference
+       function (AMDF).  The AMDF operates on low-pass, inverse filtered
+       speech.  (The low-pass filter is an 800 Hz, 19 tap, equiripple, FIR
+       filter and the inverse filter is a 2nd-order LPC filter.)  The pitch
+       estimate is later refined by dynamic tracking.  However, since some
+       of the tracking parameters are a function of the voicing decisions,
+       a voicing decision must precede the final pitch estimation. */
+    /* See subroutines LPFILT, IVFILT, and eval_highres_amdf. */
+    /* LPFILT reads indices LBUFH-LFRAME-29 = 511 through LBUFH = 720
+       of INBUF, and writes indices LBUFH+1-LFRAME = 541 through LBUFH
+       = 720 of LPBUF. */
+    lpfilt(&s->inbuf[228], &s->lpbuf[384], 312, LPC10_SAMPLES_PER_FRAME);
+    /* IVFILT reads indices (PWINH-LFRAME-7) = 353 through PWINH = 540
+       of LPBUF, and writes indices (PWINH-LFRAME+1) = 361 through
+       PWINH = 540 of IVBUF. */
+    ivfilt(&s->lpbuf[204], s->ivbuf, 312, LPC10_SAMPLES_PER_FRAME, ivrc);
+    /* eval_highres_amdf reads indices PWINL = 229 through
+       (PWINL-1)+MAXWIN+(TAU(LTAU)-TAU(1))/2 = 452 of IVBUF, and writes
+       indices 1 through LTAU = 60 of AMDF. */
+    eval_highres_amdf(s->ivbuf, 156, tau, 60, amdf, &minptr, &maxptr, &mintau);
+    /* Voicing decisions are made for each half frame of input speech.
+       An initial voicing classification is made for each half of the
+       analysis frame, and the voicing decisions for the present frame
+       are finalized.  See subroutine VOICIN. */
+    /*        The voicing detector (VOICIN) classifies the input signal as
+       unvoiced (including silence) or voiced using the AMDF windowed
+       maximum-to-minimum ratio, the zero crossing rate, energy measures,
+       reflection coefficients, and prediction gains. */
+    /*        The pitch and voicing rules apply smoothing and isolated
+       corrections to the pitch and voicing estimates and, in the process,
+       introduce two frames of delay into the corrected pitch estimates and
+       voicing decisions. */
+    for (half = 0;  half < 2;  half++)
+    {
+        lpc10_voicing(s,
+                      &s->vwin[2][0],
+                      s->inbuf,
+                      s->lpbuf,
+                      buflim,
+                      half,
+                      &amdf[minptr],
+                      &amdf[maxptr],
+                      &mintau,
+                      ivrc,
+                      s->obound);
+    }
+    /* Find the minimum cost pitch decision over several frames,
+       given the current voicing decision and the AMDF array */
+    minptr++;
+    dynamic_pitch_tracking(s, amdf, 60, &minptr, s->voibuf[3][1], pitch, &midx);
+    ipitch = tau[midx - 1];
+    /* Place spectrum analysis and energy windows */
+    lpc10_placea(&ipitch, s->voibuf, &s->obound[2], 3, s->vwin, s->awin, ewin, LPC10_SAMPLES_PER_FRAME, 156);
+    /* Remove short term DC bias over the analysis window. */
+    lanal = s->awin[2][1] + 1 - s->awin[2][0];
+    remove_dc_bias(&s->pebuf[s->awin[2][0] - 181], lanal, abuf);
+    /* Compute RMS over integer number of pitch periods within the analysis window. */
+    /* Note that in a hardware implementation this computation may be
+       simplified by using diagonal elements of phi computed by mload(). */
+    s->rmsbuf[2] = energyf(&abuf[ewin[2][0] - s->awin[2][0]], ewin[2][1] - ewin[2][0] + 1);
+    /* Matrix load and invert, check RC's for stability */
+    mload(LPC10_ORDER, 1, lanal, abuf, phi, psi);
+    invert(LPC10_ORDER, phi, psi, &s->rcbuf[2][0]);
+    rcchk(LPC10_ORDER, &s->rcbuf[1][0], &s->rcbuf[2][0]);
+    /* Set return parameters */
+    voice[0] = s->voibuf[1][0];
+    voice[1] = s->voibuf[1][1];
+    *rms = s->rmsbuf[0];
+    for (i = 0;  i < LPC10_ORDER;  i++)
+        rc[i] = s->rcbuf[0][i];
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/lpc10_decode.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/lpc10_decode.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,1106 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * lpc10_decode.c - LPC10 low bit rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the U.S. Department of Defense reference
+ * implementation of the LPC-10 2400 bps Voice Coder. They do not
+ * exert copyright claims on their code, and it may be freely used.
+ *
+ * $Id: lpc10_decode.c,v 1.16 2007/11/26 13:28:59 steveu Exp $
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <inttypes.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+#include <memory.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/lpc10.h"
+
+#define LPC10_ORDER     10
+
+#if !defined(min)
+#define min(a,b) ((a) <= (b) ? (a) : (b))
+#endif
+#if !defined(max)
+#define max(a,b) ((a) >= (b) ? (a) : (b))
+#endif
+
+/* Pseudo random number generator based on Knuth, Vol 2, p. 27. */
+/* lpc10_random - int32_t variable, uniformly distributed over -32768 to 32767 */
+static int32_t lpc10_random(lpc10_decode_state_t *s)
+{
+    int32_t ret_val;
+
+    /* The following is a 16 bit 2's complement addition,
+       with overflow checking disabled */
+    s->y[s->k] += s->y[s->j];
+    ret_val = s->y[s->k];
+    if (--s->k < 0)
+        s->k = 4;
+    if (--s->j < 0)
+        s->j = 4;
+    return ret_val;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int32_t pow_ii(int32_t x, int32_t n)
+{
+    int32_t pow;
+    uint32_t u;
+
+    if (n <= 0)
+    {
+        if (n == 0  ||  x == 1)
+            return 1;
+        if (x != -1)
+            return (x == 0)  ?  1/x  :  0;
+        n = -n;
+    }
+    u = n;
+    for (pow = 1;  ;  )
+    {
+        if ((u & 1))
+            pow *= x;
+        if ((u >>= 1) == 0)
+            break;
+        x *= x;
+    }
+    return pow;
+}
+/*- End of function --------------------------------------------------------*/
+
+/* Synthesize one pitch epoch */
+static void bsynz(lpc10_decode_state_t *s,
+                  float coef[],
+                  int32_t ip,
+                  int32_t *iv,
+                  float sout[],
+                  float rms,
+                  float ratio,
+                  float g2pass)
+{
+    static const int32_t kexc[25] =
+    {
+          8,  -16,   26, -48,  86, -162, 294, -502, 718, -728, 184, 
+        672, -610, -672, 184, 728,  718, 502,  294, 162,   86,  48, 26, 16, 8
+    };
+    int32_t i;
+    int32_t j;
+    int32_t k;
+    int32_t px;
+    float noise[166];
+    float pulse;
+    float r1;
+    float gain;
+    float xssq;
+    float sscale;
+    float xy;
+    float sum;
+    float ssq;
+    float lpi0;
+    float hpi0;
+
+    /* MAXPIT + MAXORD = 166 */
+    /* Calculate history scale factor XY and scale filter state */
+    /* Computing MIN */
+    r1 = s->rmso_bsynz/(rms + 1.0e-6f);
+    xy = min(r1, 8.0f);
+    s->rmso_bsynz = rms;
+    for (i = 0;  i < LPC10_ORDER;  i++)
+        s->exc2[i] = s->exc2[s->ipo + i]*xy;
+    s->ipo = ip;
+    if (*iv == 0)
+    {
+        /* Generate white noise for unvoiced */
+        for (i = 0;  i < ip;  i++)
+            s->exc[LPC10_ORDER + i] = (float) (lpc10_random(s)/64);
+        /* Impulse double excitation for plosives */
+        px = (lpc10_random(s) + 32768)*(ip - 1)/65536 + LPC10_ORDER + 1;
+        r1 = ratio/4.0f;
+        pulse = r1*342;
+        if (pulse > 2.0e3f)
+            pulse = 2.0e3f;
+        s->exc[px - 1] += pulse;
+        s->exc[px] -= pulse;
+    }
+    else
+    {
+        sscale = sqrtf((float) ip)/6.928f;
+        for (i = 0;  i < ip;  i++)
+        {
+            s->exc[LPC10_ORDER + i] = 0.0f;
+            if (i < 25)
+                s->exc[LPC10_ORDER + i] = sscale*kexc[i];
+            lpi0 = s->exc[LPC10_ORDER + i];
+            s->exc[LPC10_ORDER + i] = s->exc[LPC10_ORDER + i]*0.125f + s->lpi[0]*0.75f + s->lpi[1]*0.125f;
+            s->lpi[1] = s->lpi[0];
+            s->lpi[0] = lpi0;
+        }
+        for (i = 0;  i < ip;  i++)
+        {
+            noise[LPC10_ORDER + i] = lpc10_random(s)/64.0f;
+            hpi0 = noise[LPC10_ORDER + i];
+            noise[LPC10_ORDER + i] = noise[LPC10_ORDER + i]*-0.125f + s->hpi[0]*0.25f + s->hpi[1]*-0.125f;
+            s->hpi[1] = s->hpi[0];
+            s->hpi[0] = hpi0;
+        }
+        for (i = 0;  i < ip;  i++)
+            s->exc[LPC10_ORDER + i] += noise[LPC10_ORDER + i];
+    }
+    /* Synthesis filters: */
+    /* Modify the excitation with all-zero filter 1 + G*SUM */
+    xssq = 0.0f;
+    for (i = 0;  i < ip;  i++)
+    {
+        k = LPC10_ORDER + i;
+        sum = 0.0f;
+        for (j = 0;  j < LPC10_ORDER;  j++)
+            sum += coef[j]*s->exc[k - j - 1];
+        sum *= g2pass;
+        s->exc2[k] = sum + s->exc[k];
+    }
+    /* Synthesize using the all pole filter 1/(1 - SUM) */
+    for (i = 0;  i < ip;  i++)
+    {
+        k = LPC10_ORDER + i;
+        sum = 0.0f;
+        for (j = 0;  j < LPC10_ORDER;  j++)
+            sum += coef[j]*s->exc2[k - j - 1];
+        s->exc2[k] = sum + s->exc2[k];
+        xssq += s->exc2[k]*s->exc2[k];
+    }
+    /* Save filter history for next epoch */
+    for (i = 0;  i < LPC10_ORDER;  i++)
+    {
+        s->exc[i] = s->exc[ip + i];
+        s->exc2[i] = s->exc2[ip + i];
+    }
+    /* Apply gain to match RMS */
+    ssq = rms*rms*ip;
+    gain = sqrtf(ssq/xssq);
+    for (i = 0;  i < ip;  i++)
+        sout[i] = gain*s->exc2[LPC10_ORDER + i];
+}
+/*- End of function --------------------------------------------------------*/
+
+/* Synthesize a single pitch epoch */
+static int pitsyn(lpc10_decode_state_t *s,
+                  int voice[], 
+                  int32_t *pitch,
+                  float *rms,
+                  float *rc,
+                  int32_t ivuv[], 
+                  int32_t ipiti[],
+                  float *rmsi,
+                  float *rci,
+                  int32_t *nout,
+                  float *ratio)
+{
+    int32_t rci_dim1;
+    int32_t rci_offset;
+    int32_t i1;
+    int32_t i;
+    int32_t j;
+    int32_t vflag;
+    int32_t jused;
+    int32_t lsamp;
+    int32_t ip;
+    int32_t nl;
+    int32_t ivoice;
+    int32_t istart;
+    float r1;
+    float alrn;
+    float alro;
+    float yarc[10];
+    float prop;
+    float slope;
+    float uvpit;
+    float xxy;
+
+    rci_dim1 = LPC10_ORDER;
+    rci_offset = rci_dim1 + 1;
+    rci -= rci_offset;
+
+    if (*rms < 1.0f)
+        *rms = 1.0f;
+    if (s->rmso < 1.0f)
+        s->rmso = 1.0f;
+    uvpit = 0.0f;
+    *ratio = *rms/(s->rmso + 8.0f);
+    if (s->first_pitsyn)
+    {
+        lsamp = 0;
+        ivoice = voice[1];
+        if (ivoice == 0)
+            *pitch = LPC10_SAMPLES_PER_FRAME/4;
+        *nout = LPC10_SAMPLES_PER_FRAME / *pitch;
+        s->jsamp = LPC10_SAMPLES_PER_FRAME - *nout * *pitch;
+
+        i1 = *nout;
+        for (i = 0;  i < i1;  i++)
+        {
+            for (j = 0;  j < LPC10_ORDER;  j++)
+                rci[j + (i + 1)*rci_dim1 + 1] = rc[j];
+            ivuv[i] = ivoice;
+            ipiti[i] = *pitch;
+            rmsi[i] = *rms;
+        }
+        s->first_pitsyn = FALSE;
+    }
+    else
+    {
+        vflag = 0;
+        lsamp = LPC10_SAMPLES_PER_FRAME + s->jsamp;
+        slope = (*pitch - s->ipito)/(float) lsamp;
+        *nout = 0;
+        jused = 0;
+        istart = 1;
+        if (voice[0] == s->ivoico  &&  voice[1] == voice[0])
+        {
+            if (voice[1] == 0)
+            {
+                /* SSUV - -   0  ,  0  ,  0 */
+                *pitch = LPC10_SAMPLES_PER_FRAME/4;
+                s->ipito = *pitch;
+                if (*ratio > 8.0f)
+                    s->rmso = *rms;
+            }
+            /* SSVC - -   1  ,  1  ,  1 */
+            slope = (*pitch - s->ipito)/(float) lsamp;
+            ivoice = voice[1];
+        }
+        else
+        {
+            if (s->ivoico != 1)
+            {
+                if (s->ivoico == voice[0])
+                {
+                    /* UV2VC2 - -  0  ,  0  ,  1 */
+                    nl = lsamp - LPC10_SAMPLES_PER_FRAME/4;
+                }
+                else
+                {
+                    /* UV2VC1 - -  0  ,  1  ,  1 */
+                    nl = lsamp - LPC10_SAMPLES_PER_FRAME*3/4;
+                }
+                ipiti[0] = nl/2;
+                ipiti[1] = nl - ipiti[0];
+                ivuv[0] = 0;
+                ivuv[1] = 0;
+                rmsi[0] = s->rmso;
+                rmsi[1] = s->rmso;
+                for (i = 0;  i < LPC10_ORDER;  i++)
+                {
+                    rci[i + rci_dim1 + 1] = s->rco[i];
+                    rci[i + (rci_dim1 << 1) + 1] = s->rco[i];
+                    s->rco[i] = rc[i];
+                }
+                slope = 0.0f;
+                *nout = 2;
+                s->ipito = *pitch;
+                jused = nl;
+                istart = nl + 1;
+                ivoice = 1;
+            }
+            else
+            {
+                if (s->ivoico != voice[0])
+                {
+                    /* VC2UV1 - -   1  ,  0  ,  0 */
+                    lsamp = LPC10_SAMPLES_PER_FRAME/4 + s->jsamp;
+                }
+                else
+                {
+                    /* VC2UV2 - -   1  ,  1  ,  0 */
+                    lsamp = LPC10_SAMPLES_PER_FRAME*3/4 + s->jsamp;
+                }
+                for (i = 0;  i < LPC10_ORDER;  i++)
+                {
+                    yarc[i] = rc[i];
+                    rc[i] = s->rco[i];
+                }
+                ivoice = 1;
+                slope = 0.0f;
+                vflag = 1;
+            }
+        }
+        /* Here is the value of most variables that are used below, depending on */
+        /* the values of IVOICO, VOICE(1), and VOICE(2).  VOICE(1) and VOICE(2) */
+        /* are input arguments, and IVOICO is the value of VOICE(2) on the */
+        /* previous call (see notes for the IF (NOUT .NE. 0) statement near the */
+        /* end).  Each of these three values is either 0 or 1.  These three */
+        /* values below are given as 3-bit long strings, in the order IVOICO, */
+        /* VOICE(1), and VOICE(2).  It appears that the code above assumes that */
+        /* the bit sequences 010 and 101 never occur, but I wonder whether a */
+        /* large enough number of bit errors in the channel could cause such a */
+        /* thing to happen, and if so, could that cause NOUT to ever go over 11? */
+
+        /* Note that all of the 180 values in the table are floatly LFRAME, but */
+        /* 180 has fewer characters, and it makes the table a little more */
+        /* concrete.  If LFRAME is ever changed, keep this in mind.  Similarly, */
+        /* 135's are 3*LFRAME/4, and 45's are LFRAME/4.  If LFRAME is not a */
+        /* multiple of 4, then the 135 for NL-JSAMP is actually LFRAME-LFRAME/4, */
+        /* and the 45 for NL-JSAMP is actually LFRAME-3*LFRAME/4. */
+
+        /* Note that LSAMP-JSAMP is given as the variable.  This was just for */
+        /* brevity, to avoid adding "+JSAMP" to all of the column entries. */
+        /* Similarly for NL-JSAMP. */
+
+        /* Variable    | 000  001    011,010  111       110       100,101 */
+        /* ------------+-------------------------------------------------- */
+        /* ISTART      | 1    NL+1   NL+1     1         1         1 */
+        /* LSAMP-JSAMP | 180  180    180      180       135       45 */
+        /* IPITO       | 45   PITCH  PITCH    oldPITCH  oldPITCH  oldPITCH */
+        /* SLOPE       | 0    0      0        seebelow  0         0 */
+        /* JUSED       | 0    NL     NL       0         0         0 */
+        /* PITCH       | 45   PITCH  PITCH    PITCH     PITCH     PITCH */
+        /* NL-JSAMP    | --   135    45       --        --        -- */
+        /* VFLAG       | 0    0      0        0         1         1 */
+        /* NOUT        | 0    2      2        0         0         0 */
+        /* IVOICE      | 0    1      1        1         1         1 */
+
+        /* while_loop  | once once   once     once      twice     twice */
+
+        /* ISTART      | --   --     --       --        JUSED+1   JUSED+1 */
+        /* LSAMP-JSAMP | --   --     --       --        180       180 */
+        /* IPITO       | --   --     --       --        oldPITCH  oldPITCH */
+        /* SLOPE       | --   --     --       --        0         0 */
+        /* JUSED       | --   --     --       --        ??        ?? */
+        /* PITCH       | --   --     --       --        PITCH     PITCH */
+        /* NL-JSAMP    | --   --     --       --        --        -- */
+        /* VFLAG       | --   --     --       --        0         0 */
+        /* NOUT        | --   --     --       --        ??        ?? */
+        /* IVOICE      | --   --     --       --        0         0 */
+
+        /* UVPIT is always 0.0 on the first pass through the DO WHILE (TRUE)
+           loop below. */
+
+        /* The only possible non-0 value of SLOPE (in column 111) is
+           (PITCH-IPITO)/FLOAT(LSAMP) */
+
+        /* Column 101 is identical to 100.  Any good properties we can prove
+           for 100 will also hold for 101.  Similarly for 010 and 011. */
+
+        /* synths() calls this subroutine with PITCH restricted to the range 20 to
+           156.  IPITO is similarly restricted to this range, after the first
+           call.  IP below is also restricted to this range, given the
+           definitions of IPITO, SLOPE, UVPIT, and that I is in the range ISTART
+           to LSAMP. */
+
+        for (;;)
+        {
+            for (i = istart;  i <= lsamp;  i++)
+            {
+                r1 = s->ipito + slope*i;
+                ip = (int32_t) (r1 + 0.5f);
+                if (uvpit != 0.0f)
+                    ip = (int32_t) uvpit;
+                if (ip <= i - jused)
+                {
+                    ++(*nout);
+                    ipiti[*nout - 1] = ip;
+                    *pitch = ip;
+                    ivuv[*nout - 1] = ivoice;
+                    jused += ip;
+                    prop = (jused - ip/2)/(float) lsamp;
+                    for (j = 0;  j < LPC10_ORDER;  j++)
+                    {
+                        alro = logf((s->rco[j] + 1)/(1 - s->rco[j]));
+                        alrn = logf((rc[j] + 1)/(1 - rc[j]));
+                        xxy = alro + prop*(alrn - alro);
+                        xxy = expf(xxy);
+                        rci[j + *nout*rci_dim1 + 1] = (xxy - 1.0f)/(xxy + 1.0f);
+                    }
+                    rmsi[*nout - 1] = logf(s->rmso) + prop*(logf(*rms) - logf(s->rmso));
+                    rmsi[*nout - 1] = expf(rmsi[*nout - 1]);
+                }
+            }
+            if (vflag != 1)
+                break;
+
+            vflag = 0;
+            istart = jused + 1;
+            lsamp = LPC10_SAMPLES_PER_FRAME + s->jsamp;
+            slope = 0.0f;
+            ivoice = 0;
+            uvpit = (float) ((lsamp - istart)/2);
+            if (uvpit > 90.0f)
+                uvpit /= 2;
+            s->rmso = *rms;
+            for (i = 0;  i < LPC10_ORDER;  i++)
+            {
+                rc[i] = yarc[i];
+                s->rco[i] = yarc[i];
+            }
+        }
+        s->jsamp = lsamp - jused;
+    }
+    if (*nout != 0)
+    {
+        s->ivoico = voice[1];
+        s->ipito = *pitch;
+        s->rmso = *rms;
+        for (i = 0;  i < LPC10_ORDER;  i++)
+            s->rco[i] = rc[i];
+    }
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void deemp(lpc10_decode_state_t *s, float x[], int len)
+{
+    int i;
+    float r1;
+    float dei0;
+
+    for (i = 0;  i < len;  i++)
+    {
+        dei0 = x[i];
+        r1 = x[i] - s->dei[0]*1.9998f + s->dei[1];
+        x[i] = r1 + s->deo[0]*2.5f - s->deo[1]*2.0925f + s->deo[2]*0.585f;
+        s->dei[1] = s->dei[0];
+        s->dei[0] = dei0;
+        s->deo[2] = s->deo[1];
+        s->deo[1] = s->deo[0];
+        s->deo[0] = x[i];
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+/* Convert reflection coefficients to predictor coefficients */
+static float reflection_coeffs_to_predictor_coeffs(float rc[], float pc[], float gprime)
+{
+    float temp[10];
+    float g2pass;
+    int i;
+    int j;
+
+    g2pass = 1.0f;
+    for (i = 0;  i < LPC10_ORDER;  i++)
+        g2pass *= 1.0f - rc[i]*rc[i];
+    g2pass = gprime*sqrtf(g2pass);
+    pc[0] = rc[0];
+    for (i = 1;  i < LPC10_ORDER;  i++)
+    {
+        for (j = 0;  j < i;  j++)
+            temp[j] = pc[j] - rc[i]*pc[i - j - 1];
+        for (j = 0;  j < i;  j++)
+            pc[j] = temp[j];
+        pc[i] = rc[i];
+    }
+    return g2pass;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int synths(lpc10_decode_state_t *s,
+                  int voice[],
+                  int32_t *pitch,
+                  float *rms,
+                  float *rc,
+                  float speech[])
+{
+    int32_t i1;
+    int32_t ivuv[16];
+    int32_t ipiti[16];
+    int32_t nout;
+    int32_t i;
+    int32_t j;
+    float rmsi[16];
+    float ratio;
+    float g2pass;
+    float pc[10];
+    float rci[160];
+
+    i1 = min(*pitch, 156);
+    *pitch = max(i1, 20);
+    for (i = 0;  i < LPC10_ORDER;  i++)
+        rc[i] = max(min(rc[i], 0.99f), -0.99f);
+    pitsyn(s, voice, pitch, rms, rc, ivuv, ipiti, rmsi, rci, &nout, &ratio);
+    if (nout > 0)
+    {
+        for (j = 0;  j < nout;  j++)
+        {
+            /* Add synthesized speech for pitch period J to the end of s->buf. */
+            g2pass = reflection_coeffs_to_predictor_coeffs(&rci[j*10], pc, 0.7f);
+            bsynz(s, pc, ipiti[j], &ivuv[j], &s->buf[s->buflen], rmsi[j], ratio, g2pass);
+            deemp(s, &s->buf[s->buflen], ipiti[j]);
+            s->buflen += ipiti[j];
+        }
+        /* Copy first MAXFRM samples from BUF to output array speech (scaling them),
+           and then remove them from the beginning of s->buf. */
+
+        for (i = 0;  i < LPC10_SAMPLES_PER_FRAME;  i++)
+            speech[i] = s->buf[i]/4096.0f;
+        s->buflen -= LPC10_SAMPLES_PER_FRAME;
+        for (i = 0;  i < s->buflen;  i++)
+            s->buf[i] = s->buf[i + LPC10_SAMPLES_PER_FRAME];
+    }
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void lpc10_unpack(lpc10_frame_t *t, const uint8_t ibits[])
+{
+    static const int bit[10] =
+    {
+        2, 4, 8, 8, 8, 8, 16, 16, 16, 16
+    };
+    static const int iblist[53] =
+    {
+        13, 12, 11,  1,  2, 13, 12, 11,  1,  2,
+        13, 10, 11,  2,  1, 10, 13, 12, 11, 10,
+         2, 13, 12, 11, 10,  2,  1, 12,  7,  6,
+         1, 10,  9,  8,  7,  4,  6,  9,  8,  7,
+         5,  1,  9,  8,  4,  6,  1,  5,  9,  8,
+         7,  5,  6
+    };
+    int32_t itab[13];
+    int x;
+    int i;
+
+    /* ibits is 54 bits of LPC data ordered as follows: */
+    /*     R1-0, R2-0, R3-0,  P-0,  A-0, */
+    /*     R1-1, R2-1, R3-1,  P-1,  A-1, */
+    /*     R1-2, R4-0, R3-2,  A-2,  P-2, R4-1, */
+    /*     R1-3, R2-2, R3-3, R4-2,  A-3, */
+    /*     R1-4, R2-3, R3-4, R4-3,  A-4, */
+    /*      P-3, R2-4, R7-0, R8-0,  P-4, R4-4, */
+    /*     R5-0, R6-0, R7-1,R10-0, R8-1, */
+    /*     R5-1, R6-1, R7-2, R9-0,  P-5, */
+    /*     R5-2, R6-2,R10-1, R8-2,  P-6, R9-1, */
+    /*     R5-3, R6-3, R7-3, R9-2, R8-3, SYNC */
+
+    /* Reconstruct ITAB */
+    for (i = 0;  i < 13;  i++)
+        itab[i] = 0;
+    for (i = 0;  i < 53;  i++)
+    {
+        x = 52 - i;
+        x = (ibits[x >> 3] >> (7 - (x & 7))) & 1;
+        itab[iblist[52 - i] - 1] = (itab[iblist[52 - i] - 1] << 1) | x;
+    }
+    /* Sign extend the RC's */
+    for (i = 0;  i < LPC10_ORDER;  i++)
+    {
+        if ((itab[i + 3] & bit[i]))
+            itab[i + 3] -= (bit[i] << 1);
+    }
+    /* Restore variables */
+    t->ipitch = itab[0];
+    t->irms = itab[1];
+    for (i = 0;  i < LPC10_ORDER;  i++)
+        t->irc[i] = itab[LPC10_ORDER - 1 - i + 3];
+}
+/*- End of function --------------------------------------------------------*/
+
+/* Hamming 8, 4 decoder - can correct 1 out of seven bits
+   and can detect up to two errors. */
+
+/* This subroutine is entered with an eight bit word in INPUT.  The 8th */
+/* bit is parity and is stripped off.  The remaining 7 bits address the */
+/* hamming 8, 4 table and the output OUTPUT from the table gives the 4 */
+/* bits of corrected data.  If bit 4 is set,  no error was detected. */
+/* ERRCNT is the number of errors counted. */
+
+static int32_t hamming_84_decode(int32_t input, int *errcnt)
+{
+    static const uint8_t dactab[128] =
+    {
+        16,  0,  0,  3,  0,  5, 14,  7,  0,  9, 14, 11, 14, 13, 30, 14,
+         0,  9,  2,  7,  4,  7,  7, 23,  9, 25, 10,  9, 12,  9, 14,  7,
+         0,  5,  2, 11,  5, 21,  6,  5,  8, 11, 11, 27, 12,  5, 14, 11,
+         2,  1, 18,  2, 12,  5,  2,  7, 12,  9,  2, 11, 28, 12, 12, 15,
+         0,  3,  3, 19,  4, 13,  6,  3,  8, 13, 10,  3, 13, 29, 14, 13,
+         4,  1, 10,  3, 20,  4,  4,  7, 10,  9, 26, 10,  4, 13, 10, 15,
+         8,  1,  6,  3,  6,  5, 22,  6, 24,  8,  8, 11,  8, 13,  6, 15,
+         1, 17 , 2,  1,  4,  1,  6, 15,  8,  1, 10, 15, 12, 15, 15, 31
+    };
+    int i;
+    int parity;
+    int32_t output;
+
+    parity = input & 255;
+    parity ^= parity >> 4;
+    parity ^= parity >> 2;
+    parity ^= parity >> 1;
+    parity &= 1;
+    i = dactab[input & 127];
+    output = i & 15;
+    if ((i & 16))
+    {
+        /* No errors detected in seven bits */
+        if (parity)
+            (*errcnt)++;
+    }
+    else
+    {
+        /* One or two errors detected */
+        (*errcnt)++;
+        if (parity == 0)
+        {
+            /* Two errors detected */
+            (*errcnt)++;
+            output = -1;
+        }
+    }
+    return output;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int32_t median(int32_t d1, int32_t d2, int32_t d3)
+{
+    int32_t ret_val;
+
+    ret_val = d2;
+    if (d2 > d1  &&  d2 > d3)
+    {
+        ret_val = d1;
+        if (d3 > d1)
+            ret_val = d3;
+    }
+    else if (d2 < d1  &&  d2 < d3)
+    {
+        ret_val = d1;
+        if (d3 < d1)
+            ret_val = d3;
+    }
+    return ret_val;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void decode(lpc10_decode_state_t *s,
+                   lpc10_frame_t *t,
+                   int voice[],
+                   int32_t *pitch,
+                   float *rms,
+                   float rc[])
+{
+    static const int32_t ivtab[32] =
+    {
+        24960, 24960, 24960, 24960, 25480, 25480, 25483, 25480,
+        16640,  1560,  1560,  1560, 16640,  1816,  1563,  1560,
+        24960, 24960, 24859, 24856, 26001, 25881, 25915, 25913,
+         1560,  1560,  7800,  3640,  1561,  1561,  3643,  3641
+    };
+    static const float corth[32] =
+    {
+        32767.0f, 10.0f, 5.0f, 0.0f, 32767.0f,  8.0f, 4.0f, 0.0f,
+           32.0f,  6.4f, 3.2f, 0.0f,    32.0f,  6.4f, 3.2f, 0.0f,
+           32.0f, 11.2f, 6.4f, 0.0f,    32.0f, 11.2f, 6.4f, 0.0f,
+           16.0f,  5.6f, 3.2f, 0.0f,    16.0f,  5.6f, 3.2f, 0.0f
+    };
+    static const int32_t detau[128] =
+    {
+          0,   0,   0,   3,   0,   3,   3,  31,
+          0,   3,   3,  21,   3,   3,  29,  30,
+          0,   3,   3,  20,   3,  25,  27,  26,
+          3,  23,  58,  22,   3,  24,  28,   3,
+          0,   3,   3,   3,   3,  39,  33,  32,
+          3,  37,  35,  36,   3,  38,  34,   3,
+          3,  42,  46,  44,  50,  40,  48,   3,
+         54,   3,  56,   3,  52,   3,   3,   1,
+          0,   3,   3, 108,   3,  78, 100, 104,
+          3,  84,  92,  88, 156,  80,  96,   3,
+          3,  74,  70,  72,  66,  76,  68,   3,
+         62,   3,  60,   3,  64,   3,   3,   1,
+          3, 116, 132, 112, 148, 152,   3,   3,
+        140,   3, 136,   3, 144,   3,   3,   1,
+        124, 120, 128,   3,   3,   3,   3,   1,
+          3,   3,   3,   1,   3,   1,   1,   1
+    };
+    static const int32_t rmst[64] =
+    {
+        1024,  936,  856,  784,  718,  656,  600,  550,
+         502,  460,  420,  384,  352,  328,  294,  270,
+         246,  226,  206,  188,  172,  158,  144,  132,
+         120,  110,  102,   92,   84,   78,   70,   64,
+          60,   54,   50,   46,   42,   38,   34,   32,
+          30,   26,   24,   22,   20,   18,   17,   16,
+          15,   14,   13,   12,   11,   10,    9,    8,
+           7,    6,    5,    4,    3,    2,    1,    0
+    };
+    static const int32_t detab7[32] =
+    {
+          4,  11,  18,  25,  32,  39,  46,  53,
+         60,  66,  72,  77,  82,  87,  92,  96,
+        101, 104, 108, 111, 114, 115, 117, 119,
+        121, 122, 123, 124, 125, 126, 127, 127
+    };
+    static const float descl[8] =
+    {
+        0.6953f, 0.625f, 0.5781f, 0.5469f, 0.5312f, 0.5391f, 0.4688f, 0.3828f
+    };
+    static const int32_t deadd[8] =
+    {
+        1152, -2816, -1536, -3584, -1280, -2432, 768, -1920
+    };
+    static const int32_t qb[8] =
+    {
+        511, 511, 1023, 1023, 1023, 1023, 2047, 4095
+    };
+    static const int32_t nbit[10] =
+    {
+        8, 8, 5, 5, 4, 4, 4, 4, 3, 2
+    };
+    static const int32_t zrc[10] =
+    {
+        0, 0, 0, 0, 0, 3, 0, 2, 0, 0
+    };
+    static const int32_t bit[5] =
+    {
+        2, 4, 8, 16, 32
+    };
+    int32_t ipit;
+    int32_t iout;
+    int32_t i;
+    int32_t icorf;
+    int32_t index;
+    int32_t ivoic;
+    int32_t ixcor;
+    int32_t i1;
+    int32_t i2;
+    int32_t i4;
+    int32_t ishift;
+    int32_t lsb;
+    int errcnt;
+
+    /* If no error correction, do pitch and voicing then jump to decode */
+    i4 = detau[t->ipitch];
+    if (!s->error_correction)
+    {
+        voice[0] = 1;
+        voice[1] = 1;
+        if (t->ipitch <= 1)
+            voice[0] = 0;
+        if (t->ipitch == 0  ||  t->ipitch == 2)
+            voice[1] = 0;
+        if (i4 <= 4)
+            i4 = s->iptold;
+        *pitch = i4;
+        if (voice[0] == 1  &&  voice[1] == 1)
+            s->iptold = *pitch;
+        if (voice[0] != voice[1])
+            *pitch = s->iptold;
+    }
+    else
+    {
+        /* Do error correction pitch and voicing */
+        if (i4 > 4)
+        {
+            s->dpit[0] = i4;
+            ivoic = 2;
+            s->iavgp = (s->iavgp*15 + i4 + 8)/16;
+        }
+        else
+        {
+            s->dpit[0] = s->iavgp;
+            ivoic = i4;
+        }
+        s->drms[0] = t->irms;
+        for (i = 0;  i < LPC10_ORDER;  i++)
+            s->drc[i][0] = t->irc[i];
+        /* Determine index to IVTAB from V/UV decision */
+        /* If error rate is high then use alternate table */
+        index = (s->ivp2h << 4) + (s->iovoic << 2) + ivoic + 1;
+        i1 = ivtab[index - 1];
+        ipit = i1 & 3;
+        icorf = i1 >> 3;
+        if (s->erate < 2048)
+            icorf /= 64;
+        /* Determine error rate:  4=high    1=low */
+        ixcor = 4;
+        if (s->erate < 2048)
+            ixcor = 3;
+        if (s->erate < 1024)
+            ixcor = 2;
+        if (s->erate < 128)
+            ixcor = 1;
+        /* Voice/unvoice decision determined from bits 0 and 1 of IVTAB */
+        voice[0] = icorf/2 & 1;
+        voice[1] = icorf & 1;
+        /* Skip decoding on first frame because present data not yet available */
+        if (s->first)
+        {
+            s->first = FALSE;
+            /* Assign PITCH a "default" value on the first call, since */
+            /* otherwise it would be left uninitialized.  The two lines */
+            /* below were copied from above, since it seemed like a */
+            /* reasonable thing to do for the first call. */
+            if (i4 <= 4)
+                i4 = s->iptold;
+            *pitch = i4;
+        }
+        else
+        {
+            /* If bit 4 of ICORF is set then correct RMS and RC(1) - RC(4). */
+            /* Determine error rate and correct errors using a Hamming 8,4 code */
+            /* during transition of unvoiced frames.  If IOUT is negative, */
+            /* more than 1 error occurred, use previous frame's parameters. */
+            if ((icorf & bit[3]) != 0)
+            {
+                errcnt = 0;
+                lsb = s->drms[1] & 1;
+                index = (s->drc[7][1] << 4) + s->drms[1]/2;
+                iout = hamming_84_decode(index, &errcnt);
+                s->drms[1] = s->drms[2];
+                if (iout >= 0)
+                    s->drms[1] = (iout << 1) + lsb;
+                for (i = 1;  i <= 4;  i++)
+                {
+                    if (i == 1)
+                        i1 = ((s->drc[8][1] & 7) << 1) + (s->drc[9][1] & 1);
+                    else
+                        i1 = s->drc[8 - i][1] & 15;
+                    i2 = s->drc[4 - i][1] & 31;
+                    lsb = i2 & 1;
+                    index = (i1 << 4) + (i2 >> 1);
+                    iout = hamming_84_decode(index, &errcnt);
+                    if (iout >= 0)
+                    {
+                        iout = (iout << 1) + lsb;
+                        if ((iout & 16) == 16)
+                            iout -= 32;
+                    }
+                    else
+                    {
+                        iout = s->drc[4 - i][2];
+                    }
+                    s->drc[4 - i][1] = iout;
+                }
+                /* Determine error rate */
+                s->erate = (int32_t) (s->erate*0.96875f + errcnt*102.0f);
+            }
+            /* Get unsmoothed RMS, RC's, and PITCH */
+            t->irms = s->drms[1];
+            for (i = 0;  i < LPC10_ORDER;  i++)
+                t->irc[i] = s->drc[i][1];
+            if (ipit == 1)
+                s->dpit[1] = s->dpit[2];
+            if (ipit == 3)
+                s->dpit[1] = s->dpit[0];
+            *pitch = s->dpit[1];
+            /* If bit 2 of ICORF is set then smooth RMS and RC's, */
+            if ((icorf & bit[1]) != 0)
+            {
+                if ((float) abs(s->drms[1] - s->drms[0]) >= corth[ixcor + 3] 
+                    &&
+                    (float) abs(s->drms[1] - s->drms[2]) >= corth[ixcor + 3])
+                {
+                    t->irms = median(s->drms[2], s->drms[1], s->drms[0]);
+                }
+                for (i = 0;  i < 6;  i++)
+                {
+                    if ((float) abs(s->drc[i][1] - s->drc[i][0]) >= corth[ixcor + ((i + 3) << 2) - 5]
+                        &&
+                        (float) abs(s->drc[i][1] - s->drc[i][2]) >= corth[ixcor + ((i + 3) << 2) - 5])
+                    {
+                        t->irc[i] = median(s->drc[i][2], s->drc[i][1], s->drc[i][0]);
+                    }
+                }
+            }
+            /* If bit 3 of ICORF is set then smooth pitch */
+            if ((icorf & bit[2]) != 0)
+            {
+                if ((float) abs(s->dpit[1] - s->dpit[0]) >= corth[ixcor - 1] 
+                    &&
+                    (float) abs(s->dpit[1] - s->dpit[2]) >= corth[ixcor - 1])
+                {
+                    *pitch = median(s->dpit[2], s->dpit[1], s->dpit[0]);
+                }
+            }
+            /* If bit 5 of ICORF is set then RC(5) - RC(10) are loaded with
+               values so that after quantization bias is removed in decode
+               the values will be zero. */
+        }
+        if ((icorf & bit[4]) != 0)
+        {
+            for (i = 4;  i < LPC10_ORDER;  i++)
+                t->irc[i] = zrc[i];
+        }
+        /* Housekeeping  - one frame delay */
+        s->iovoic = ivoic;
+        s->ivp2h = voice[1];
+        s->dpit[2] = s->dpit[1];
+        s->dpit[1] = s->dpit[0];
+        s->drms[2] = s->drms[1];
+        s->drms[1] = s->drms[0];
+        for (i = 0;  i < LPC10_ORDER;  i++)
+        {
+            s->drc[i][2] = s->drc[i][1];
+            s->drc[i][1] = s->drc[i][0];
+        }
+    }
+    /* Decode RMS */
+    t->irms = rmst[(31 - t->irms)*2];
+    /* Decode RC(1) and RC(2) from log-area-ratios */
+    /* Protect from illegal coded value (-16) caused by bit errors */
+    for (i = 0;  i < 2;  i++)
+    {
+        i2 = t->irc[i];
+        i1 = 0;
+        if (i2 < 0)
+        {
+            i1 = 1;
+            i2 = -i2;
+            if (i2 > 15)
+                i2 = 0;
+        }
+        i2 = detab7[i2*2];
+        if (i1 == 1)
+            i2 = -i2;
+        ishift = 15 - nbit[i];
+        t->irc[i] = i2*pow_ii(2, ishift);
+    }
+    /* Decode RC(3)-RC(10) to sign plus 14 bits */
+    for (i = 2;  i < LPC10_ORDER;  i++)
+    {
+        ishift = 15 - nbit[i];
+        i2 = t->irc[i]*pow_ii(2, ishift) + qb[i - 2];
+        t->irc[i] = (int32_t) (i2*descl[i - 2] + deadd[i - 2]);
+    }
+    /* Scale RMS and RC's to floats */
+    *rms = (float) t->irms;
+    for (i = 0;  i < LPC10_ORDER;  i++)
+        rc[i] = t->irc[i]/16384.0f;
+}
+/*- End of function --------------------------------------------------------*/
+
+lpc10_decode_state_t *lpc10_decode_init(lpc10_decode_state_t *s, int error_correction)
+{
+    static const int16_t rand_init[] =
+    {
+        -21161,
+         -8478,
+         30892,
+        -10216,
+         16950
+    };
+    int i;
+    int j;
+
+    if (s == NULL)
+    {
+        if ((s = (lpc10_decode_state_t *) malloc(sizeof(*s))) == NULL)
+            return NULL;
+    }
+
+    s->error_correction = error_correction;
+
+    /* State used by function decode */
+    s->iptold = 60;
+    s->first = TRUE;
+    s->ivp2h = 0;
+    s->iovoic = 0;
+    s->iavgp = 60;
+    s->erate = 0;
+    for (i = 0;  i < 3;  i++)
+    {
+        for (j = 0;  j < 10;  j++)
+            s->drc[j][i] = 0;
+        s->dpit[i] = 0;
+        s->drms[i] = 0;
+    }
+
+    /* State used by function synths */
+    for (i = 0;  i < 360;  i++)
+        s->buf[i] = 0.0f;
+    s->buflen = LPC10_SAMPLES_PER_FRAME;
+
+    /* State used by function pitsyn */
+    s->rmso = 1.0f;
+    s->first_pitsyn = TRUE;
+
+    /* State used by function bsynz */
+    s->ipo = 0;
+    for (i = 0;  i < 166;  i++)
+    {
+        s->exc[i] = 0.0f;
+        s->exc2[i] = 0.0f;
+    }
+    for (i = 0;  i < 3;  i++)
+    {
+        s->lpi[i] = 0.0f;
+        s->hpi[i] = 0.0f;
+    }
+    s->rmso_bsynz = 0.0f;
+
+    /* State used by function lpc10_random */
+    s->j = 1;
+    s->k = 4;
+    for (i = 0;  i < 5;  i++)
+        s->y[i] = rand_init[i];
+
+    /* State used by function deemp */
+    for (i = 0;  i < 2;  i++)
+        s->dei[i] = 0.0f;
+    for (i = 0;  i < 3;  i++)
+        s->deo[i] = 0.0f;
+    
+    return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int lpc10_decode_release(lpc10_decode_state_t *s)
+{
+    free(s);
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+int lpc10_decode(lpc10_decode_state_t *s, int16_t amp[], const uint8_t code[], int quant)
+{
+    int voice[2];
+    int32_t pitch;
+    float speech[LPC10_SAMPLES_PER_FRAME];
+    float rc[LPC10_ORDER];
+    lpc10_frame_t frame;
+    float rms;
+    int i;
+    int j;
+
+    /* Decode 54 bits to LPC10_SAMPLES_PER_FRAME speech samples. */
+    for (i = 0;  i < quant;  i++)
+    {
+        lpc10_unpack(&frame, &code[i*7]);
+        decode(s, &frame, voice, &pitch, &rms, rc);
+        synths(s, voice, &pitch, &rms, rc, speech);
+        for (j = 0;  j < LPC10_SAMPLES_PER_FRAME;  j++)
+            amp[i*LPC10_SAMPLES_PER_FRAME + j] = (int16_t) rintf(32768.0f*speech[j]);
+    }
+
+    return quant*LPC10_SAMPLES_PER_FRAME;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/lpc10_encdecs.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/lpc10_encdecs.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,107 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * lpc10_encdecs.h - LPC10 low bit rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: lpc10_encdecs.h,v 1.9 2006/11/30 15:41:47 steveu Exp $
+ */
+
+#define LPC10_ORDER     10
+
+#if !defined(min)
+#define min(a,b) ((a) <= (b) ? (a) : (b))
+#endif
+#if !defined(max)
+#define max(a,b) ((a) >= (b) ? (a) : (b))
+#endif
+
+void lpc10_placea(int32_t *ipitch,
+                  int32_t voibuf[4][2],
+                  int32_t *obound,
+                  int32_t af,
+                  int32_t vwin[3][2],
+                  int32_t awin[3][2],
+                  int32_t ewin[3][2], 
+                  int32_t lframe,
+                  int32_t maxwin);
+
+void lpc10_placev(int32_t *osbuf,
+                  int32_t *osptr,
+                  int32_t oslen, 
+                  int32_t *obound,
+                  int32_t vwin[3][2],
+                  int32_t af,
+                  int32_t lframe,
+                  int32_t minwin,
+                  int32_t maxwin,
+                  int32_t dvwinl,
+                  int32_t dvwinh);
+
+void lpc10_voicing(lpc10_encode_state_t *st,
+                   int32_t *vwin,
+                   float *inbuf,
+                   float *lpbuf,
+                   const int32_t buflim[],
+                   int32_t half,
+                   float *minamd,
+                   float *maxamd, 
+	               int32_t *mintau,
+                   float *ivrc,
+                   int32_t *obound);
+
+void lpc10_analyse(lpc10_encode_state_t *st, float *speech, int32_t *voice, int32_t *pitch, float *rms, float rc[]);
+
+static __inline__ int32_t pow_ii(int32_t x, int32_t n)
+{
+    int32_t pow;
+    uint32_t u;
+
+    if (n <= 0)
+    {
+        if (n == 0  ||  x == 1)
+            return 1;
+        if (x != -1)
+            return (x == 0)  ?  1/x  :  0;
+        n = -n;
+    }
+    u = n;
+    for (pow = 1;  ;  )
+    {
+        if ((u & 1))
+            pow *= x;
+        if ((u >>= 1) == 0)
+            break;
+        x *= x;
+    }
+    return pow;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ float r_sign(float a, float b)
+{
+    float x;
+
+    x = fabsf(a);
+    return (b >= 0.0f)  ?  x  :  -x;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/lpc10_encode.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/lpc10_encode.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,394 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * lpc10_encode.c - LPC10 low bit rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the U.S. Department of Defense reference
+ * implementation of the LPC-10 2400 bps Voice Coder. They do not
+ * exert copyright claims on their code, and it may be freely used.
+ *
+ * $Id: lpc10_encode.c,v 1.17 2007/11/26 13:28:59 steveu Exp $
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <inttypes.h>
+#include <memory.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/lpc10.h"
+
+#include "lpc10_encdecs.h"
+
+static void lpc10_pack(lpc10_encode_state_t *s, uint8_t ibits[], lpc10_frame_t *t)
+{
+    static const int iblist[53] =
+    {
+        13, 12, 11,  1,  2, 13, 12, 11,  1,  2,
+        13, 10, 11,  2,  1, 10, 13, 12, 11, 10,
+         2, 13, 12, 11, 10,  2,  1, 12,  7,  6,
+         1, 10,  9,  8,  7,  4,  6,  9,  8,  7,
+         5,  1,  9,  8,  4,  6,  1,  5,  9,  8,
+         7,  5,  6
+    };
+    int32_t itab[13];
+    int x;
+    int i;
+
+    /* ibits is 54 bits of LPC data ordered as follows: */
+    /*     R1-0, R2-0, R3-0,  P-0,  A-0, */
+    /*     R1-1, R2-1, R3-1,  P-1,  A-1, */
+    /*     R1-2, R4-0, R3-2,  A-2,  P-2, R4-1, */
+    /*     R1-3, R2-2, R3-3, R4-2,  A-3, */
+    /*     R1-4, R2-3, R3-4, R4-3,  A-4, */
+    /*      P-3, R2-4, R7-0, R8-0,  P-4, R4-4, */
+    /*     R5-0, R6-0, R7-1,R10-0, R8-1, */
+    /*     R5-1, R6-1, R7-2, R9-0,  P-5, */
+    /*     R5-2, R6-2,R10-1, R8-2,  P-6, R9-1, */
+    /*     R5-3, R6-3, R7-3, R9-2, R8-3, SYNC */
+
+    itab[0] = t->ipitch;
+    itab[1] = t->irms;
+    itab[2] = 0;
+    for (i = 0;  i < LPC10_ORDER;  i++)
+        itab[i + 3] = t->irc[LPC10_ORDER - 1 - i] & 0x7FFF;
+    /* Put 54 bits into the output buffer */
+    x = 0;
+    for (i = 0;  i < 53;  i++)
+    {
+        x = (x << 1) | (itab[iblist[i] - 1] & 1);
+        if ((i & 7) == 7)
+            ibits[i >> 3] = (uint8_t) (x & 0xFF);
+        itab[iblist[i] - 1] >>= 1;
+    }
+    x = (x << 1) | (s->isync & 1);
+    s->isync ^= 1;
+    x <<= 2;
+    ibits[6] = (uint8_t) (x & 0xFF);
+}
+/*- End of function --------------------------------------------------------*/
+
+/* Quantize LPC parameters for transmission */
+static int encode(lpc10_encode_state_t *s,
+                  lpc10_frame_t *t,
+                  int32_t *voice,
+                  int32_t pitch,
+                  float rms,
+                  float *rc)
+{
+    static const int32_t enctab[16] =
+    {
+        0, 7, 11, 12, 13, 10, 6, 1, 14, 9, 5, 2, 3, 4, 8, 15
+    };
+    static const int32_t entau[60] =
+    {
+        19, 11, 27, 25, 29, 21, 23, 22, 30, 14, 15,  7, 39, 38, 46, 
+        42, 43, 41, 45, 37, 53, 49, 51, 50, 54, 52, 60, 56, 58, 26,
+        90, 88, 92, 84, 86, 82, 83, 81, 85, 69, 77, 73, 75, 74, 78,
+        70, 71, 67, 99, 97, 113, 112, 114, 98, 106, 104, 108, 100,
+        101, 76
+    };
+    static const int32_t enadd[8] =
+    {
+        1920, -768, 2432, 1280, 3584, 1536, 2816, -1152
+    };
+    static const float enscl[8] =
+    {
+        0.0204f, 0.0167f, 0.0145f, 0.0147f, 0.0143f, 0.0135f, 0.0125f, 0.0112f
+    };
+    static const int32_t enbits[8] =
+    {
+        6, 5, 4, 4, 4, 4, 3, 3
+    };
+    static const int32_t entab6[64] =
+    {
+        0, 0, 0, 0, 0, 0, 1, 1, 1, 1, 1, 1, 1, 2, 2, 2, 2, 2, 2, 2, 3, 3,
+        3, 3, 3, 3, 3, 4, 4, 4, 4, 4, 4, 4, 5, 5, 5, 5, 5, 6, 6, 6, 6, 6,
+        7, 7, 7, 7, 7, 8, 8, 8, 8, 9, 9, 9, 10, 10, 11, 11, 12, 13, 14, 15
+    };
+    static const int32_t rmst[64] =
+    {
+        1024, 936, 856, 784, 718, 656, 600, 550, 502,
+        460, 420, 384, 352, 328, 294, 270, 246, 226,
+        206, 188, 172, 158, 144, 132, 120, 110, 102, 
+        92, 84, 78, 70, 64, 60, 54, 50,
+        46, 42, 38, 34, 32, 30, 26, 24,
+        22, 20, 18, 17, 16, 15, 14, 13,
+        12, 11, 10, 9, 8, 7, 6, 5, 4,
+        3, 2, 1, 0
+    };
+
+    int32_t idel;
+    int32_t nbit;
+    int32_t i;
+    int32_t j;
+    int32_t i2;
+    int32_t i3;
+    int32_t mrk;
+
+    /* Scale RMS and RC's to int32_ts */
+    t->irms = (int32_t) rms;
+    for (i = 0;  i < LPC10_ORDER;  i++)
+        t->irc[i] = (int32_t) (rc[i]*32768.0f);
+    if (voice[0] != 0  &&  voice[1] != 0)
+    {
+        t->ipitch = entau[pitch - 1];
+    }
+    else
+    {
+        if (s->error_correction)
+        {
+            t->ipitch = 0;
+            if (voice[0] != voice[1])
+                t->ipitch = 127;
+        }
+        else
+        {
+            t->ipitch = (voice[0] << 1) + voice[1];
+        }
+    }
+    /* Encode RMS by binary table search */
+    j = 32;
+    idel = 16;
+    t->irms = min(t->irms, 1023);
+    while (idel > 0)
+    {
+        if (t->irms > rmst[j - 1])
+            j -= idel;
+        if (t->irms < rmst[j - 1])
+            j += idel;
+        idel /= 2;
+    }
+    if (t->irms > rmst[j - 1])
+        --j;
+    t->irms = 31 - j/2;
+    /* Encode RC(1) and (2) as log-area-ratios */
+    for (i = 0;  i < 2;  i++)
+    {
+        i2 = t->irc[i];
+        mrk = 0;
+        if (i2 < 0)
+        {
+            i2 = -i2;
+            mrk = 1;
+        }
+        i2 = min(i2/512, 63);
+        i2 = entab6[i2];
+        if (mrk != 0)
+            i2 = -i2;
+        t->irc[i] = i2;
+    }
+    /* Encode RC(3) - (10) linearly, remove bias then scale */
+    for (i = 2;  i < LPC10_ORDER;  i++)
+    {
+        i2 = (int32_t) ((t->irc[i]/2 + enadd[LPC10_ORDER - 1 - i])*enscl[LPC10_ORDER - 1 - i]);
+        i2 = max(i2, -127);
+        i2 = min(i2, 127);
+        nbit = enbits[LPC10_ORDER - 1 - i];
+        i3 = (i2 < 0);
+        i2 /= pow_ii(2, nbit);
+        if (i3)
+            i2--;
+        t->irc[i] = i2;
+    }
+    /* Protect the most significant bits of the most
+       important parameters during non-voiced frames.
+       RC(1) - RC(4) are protected using 20 parity bits
+       replacing RC(5) - RC(10). */
+    if (s->error_correction)
+    {
+        if (t->ipitch == 0  ||  t->ipitch == 127)
+        {
+            t->irc[4] = enctab[(t->irc[0] & 0x1E) >> 1];
+            t->irc[5] = enctab[(t->irc[1] & 0x1E) >> 1];
+            t->irc[6] = enctab[(t->irc[2] & 0x1E) >> 1];
+            t->irc[7] = enctab[(t->irms & 0x1E) >> 1];
+            t->irc[8] = enctab[(t->irc[3] & 0x1E) >> 1] >> 1;
+            t->irc[9] = enctab[(t->irc[3] & 0x1E) >> 1] & 1;
+        }
+    }
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+static void high_pass_100hz(lpc10_encode_state_t *s, float speech[], int start, int len)
+{
+    float si;
+    float err;
+    int i;
+
+    /* 100 Hz high pass filter */
+    for (i = start;  i < len;  i++)
+    {
+        si = speech[i];
+        err = si + s->z11*1.859076f - s->z21*0.8648249f;
+        si = err - s->z11*2.0f + s->z21;
+        s->z21 = s->z11;
+        s->z11 = err;
+        err = si + s->z12*1.935715f - s->z22*0.9417004f;
+        si = err - s->z12*2.0f + s->z22;
+        s->z22 = s->z12;
+        s->z12 = err;
+        speech[i] = si*0.902428f;
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+lpc10_encode_state_t *lpc10_encode_init(lpc10_encode_state_t *s, int error_correction)
+{
+    int i;
+    int j;
+
+    if (s == NULL)
+    {
+        if ((s = (lpc10_encode_state_t *) malloc(sizeof(*s))) == NULL)
+            return NULL;
+    }
+
+    s->error_correction = error_correction;
+
+    /* State used only by function high_pass_100hz */
+    s->z11 = 0.0f;
+    s->z21 = 0.0f;
+    s->z12 = 0.0f;
+    s->z22 = 0.0f;
+    
+    /* State used by function lpc10_analyse */
+    for (i = 0;  i < 540;  i++)
+    {
+        s->inbuf[i] = 0.0f;
+        s->pebuf[i] = 0.0f;
+    }
+    for (i = 0;  i < 696;  i++)
+        s->lpbuf[i] = 0.0f;
+    for (i = 0;  i < 312;  i++)
+        s->ivbuf[i] = 0.0f;
+    s->bias = 0.0f;
+    s->osptr = 1;
+    for (i = 0;  i < 3;  i++)
+        s->obound[i] = 0;
+    s->vwin[2][0] = 307;
+    s->vwin[2][1] = 462;
+    s->awin[2][0] = 307;
+    s->awin[2][1] = 462;
+    for (i = 0;  i < 4;  i++)
+    {
+        s->voibuf[i][0] = 0;
+        s->voibuf[i][1] = 0;
+    }
+    for (i = 0;  i < 3;  i++)
+        s->rmsbuf[i] = 0.0f;
+    for (i = 0;  i < 3;  i++)
+    {
+        for (j = 0;  j < 10;  j++)
+            s->rcbuf[i][j] = 0.0f;
+    }
+    s->zpre = 0.0f;
+
+    /* State used by function onset */
+    s->n = 0.0f;
+    s->d__ = 1.0f;
+    for (i = 0;  i < 16;  i++)
+        s->l2buf[i] = 0.0f;
+    s->l2sum1 = 0.0f;
+    s->l2ptr1 = 1;
+    s->l2ptr2 = 9;
+    s->hyst = FALSE;
+
+    /* State used by function lpc10_voicing */
+    s->dither = 20.0f;
+    s->maxmin = 0.0f;
+    for (i = 0;  i < 3;  i++)
+    {
+        s->voice[i][0] = 0.0f;
+        s->voice[i][1] = 0.0f;
+    }
+    s->lbve = 3000;
+    s->fbve = 3000;
+    s->fbue = 187;
+    s->ofbue = 187;
+    s->sfbue = 187;
+    s->lbue = 93;
+    s->olbue = 93;
+    s->slbue = 93;
+    s->snr = (float) (s->fbve / s->fbue << 6);
+
+    /* State used by function dynamic_pitch_tracking */
+    for (i = 0;  i < 60;  i++)
+        s->s[i] = 0.0f;
+    for (i = 0;  i < 2;  i++)
+    {
+        for (j = 0;  j < 60;  j++)
+            s->p[i][j] = 0;
+    }
+    s->ipoint = 0;
+    s->alphax = 0.0f;
+
+    /* State used by function lpc10_pack */
+    s->isync = 0;
+    
+    return s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int lpc10_encode_release(lpc10_encode_state_t *s)
+{
+    free(s);
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+int lpc10_encode(lpc10_encode_state_t *s, uint8_t code[], const int16_t amp[], int quant)
+{
+    int32_t voice[2];
+    int32_t pitch;
+    float speech[LPC10_SAMPLES_PER_FRAME];
+    float rc[LPC10_ORDER];
+    float rms;
+    lpc10_frame_t frame;
+    int i;
+    int j;
+
+    for (i = 0;  i < quant;  i++)
+    {
+        for (j = 0;  j < LPC10_SAMPLES_PER_FRAME;  j++)
+            speech[j] = (float) amp[i*LPC10_SAMPLES_PER_FRAME + j]/32768.0f;
+        high_pass_100hz(s, speech, 0, LPC10_SAMPLES_PER_FRAME);
+        lpc10_analyse(s, speech, voice, &pitch, &rms, rc);
+        encode(s, &frame, voice, pitch, rms, rc);
+        lpc10_pack(s, &code[7*i], &frame);
+    }
+    return quant*7;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/lpc10_placev.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/lpc10_placev.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,335 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * lpc10_placev.c - LPC10 low bit rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the U.S. Department of Defense reference
+ * implementation of the LPC-10 2400 bps Voice Coder. They do not
+ * exert copyright claims on their code, and it may be freely used.
+ *
+ * $Id: lpc10_placev.c,v 1.12 2007/01/03 14:15:35 steveu Exp $
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <inttypes.h>
+#include <memory.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/lpc10.h"
+
+#include "lpc10_encdecs.h"
+
+#define subsc(x,y) (((x) << 1) + (y))
+
+void lpc10_placea(int32_t *ipitch,
+                  int32_t voibuf[3][2],
+                  int32_t *obound,
+                  int32_t af,
+                  int32_t vwin[3][2],
+                  int32_t awin[3][2],
+                  int32_t ewin[3][2], 
+                  int32_t lframe,
+                  int32_t maxwin)
+{
+    int allv;
+    int winv;
+    int32_t i;
+    int32_t j;
+    int32_t k;
+    int32_t l;
+    int32_t hrange;
+    int ephase;
+    int32_t lrange;
+    
+    lrange = (af - 2)*lframe + 1;
+    hrange = af*lframe;
+
+    /* Place the analysis window based on the voicing window placement,
+       onsets, tentative voicing decision, and pitch. */
+
+    /* Case 1:  Sustained voiced speech
+       If the five most recent voicing decisions are
+       voiced, then the window is placed phase-synchronously with the
+       previous window, as close to the present voicing window if possible.
+       If onsets bound the voicing window, then preference is given to
+       a phase-synchronous placement which does not overlap these onsets. */
+
+    /* Case 2:  Voiced transition
+       If at least one voicing decision in AF is voicied, and there are no
+       onsets, then the window is placed as in case 1. */
+
+    /* Case 3:  Unvoiced speech or onsets
+       If both voicing decisions in AF are unvoiced, or there are onsets
+       then the window is placed coincident with the voicing window. */
+
+    /* Note:  During phase-synchronous placement of windows, the length
+       is not altered from MAXWIN, since this would defeat the purpose
+       of phase-synchronous placement. */
+
+    /* Check for case 1 and case 2 */
+    allv = voibuf[af - 2][1] == 1
+           &&
+           voibuf[af - 1][0] == 1
+           &&
+           voibuf[af - 1][1] == 1
+           &&
+           voibuf[af][0] == 1
+           &&
+           voibuf[af][1] == 1;
+    winv = voibuf[af][0] == 1  ||  voibuf[af][1] == 1;
+    if (allv  ||  (winv  &&  *obound == 0))
+    {
+        /* APHASE:  Phase synchronous window placement. */
+        /* Get minimum lower index of the window. */
+        i = (lrange + *ipitch - 1 - awin[af - 2][0]) / *ipitch;
+        i *= *ipitch;
+        i += awin[af - 2][0];
+        /* l = the actual length of this frame's analysis window. */
+        l = maxwin;
+        /* Calculate the location where a perfectly centered window would start. */
+        k = (vwin[af - 1][0] + vwin[af - 1][1] + 1 - l)/2;
+        /* Choose the actual location to be the pitch multiple closest to this */
+        awin[af - 1][0] = i + ((int) floorf((float) (k - i)/(float) *ipitch + 0.5f))*(*ipitch);
+        awin[af - 1][1] = awin[af - 1][0] + l - 1;
+        /* If there is an onset bounding the right of the voicing window and the
+           analysis window overlaps that, then move the analysis window backward
+           to avoid this onset. */
+        if (*obound >= 2  &&  awin[af - 1][1] > vwin[af - 1][1])
+        {
+            awin[af - 1][0] -= *ipitch;
+            awin[af - 1][1] -= *ipitch;
+        }
+        /* Similarly for the left of the voicing window. */
+        if ((*obound == 1  ||  *obound == 3)  &&  awin[af - 1][0] < vwin[af - 1][0])
+        {
+            awin[af - 1][0] += *ipitch;
+            awin[af - 1][1] += *ipitch;
+        }
+        /* If this placement puts the analysis window above HRANGE, then
+           move it backward an integer number of pitch periods. */
+        while (awin[af - 1][1] > hrange)
+        {
+            awin[af - 1][0] -= *ipitch;
+            awin[af - 1][1] -= *ipitch;
+        }
+        /* Similarly if the placement puts the analysis window below LRANGE. */
+        while (awin[af - 1][0] < lrange)
+        {
+            awin[af - 1][0] += *ipitch;
+            awin[af - 1][1] += *ipitch;
+        }
+        /* Make energy window be phase-synchronous. */
+        ephase = TRUE;
+    }
+    else
+    {
+        /* Case 3 */
+        awin[af - 1][0] = vwin[af - 1][0];
+        awin[af - 1][1] = vwin[af - 1][1];
+        ephase = FALSE;
+    }
+    /* RMS is computed over an integer number of pitch periods in the analysis
+       window.  When it is not placed phase-synchronously, it is placed as close
+       as possible to onsets. */
+    j = (awin[af - 1][1] - awin[af - 1][0] + 1) / *ipitch * *ipitch;
+    if (j == 0  ||  !winv)
+    {
+        ewin[af - 1][0] = vwin[af - 1][0];
+        ewin[af - 1][1] = vwin[af - 1][1];
+    }
+    else if (!ephase  &&  *obound == 2)
+    {
+        ewin[af - 1][0] = awin[af - 1][1] - j + 1;
+        ewin[af - 1][1] = awin[af - 1][1];
+    }
+    else
+    {
+        ewin[af - 1][0] = awin[af - 1][0];
+        ewin[af - 1][1] = awin[af - 1][0] + j - 1;
+    }
+}
+/*- End of function --------------------------------------------------------*/
+
+void lpc10_placev(int32_t *osbuf,
+                  int32_t *osptr,
+                  int32_t oslen, 
+                  int32_t *obound,
+                  int32_t vwin[3][2],
+                  int32_t af,
+                  int32_t lframe,
+                  int32_t minwin,
+                  int32_t maxwin,
+                  int32_t dvwinl,
+                  int32_t dvwinh)
+{
+    int32_t i1;
+    int32_t i2;
+    int crit;
+    int32_t q;
+    int32_t osptr1;
+    int32_t hrange;
+    int32_t lrange;
+    int i;
+
+    /* Voicing window placement */
+
+    /*         __________________ __________________ ______________ */
+    /*        |                  |                  |               */
+    /*        |        1F        |        2F        |        3F ... */
+    /*        |__________________|__________________|______________ */
+
+    /*    Previous | */
+    /*      Window | */
+    /*  ...________| */
+
+    /*             |                                | */
+    /*      ------>| This window's placement range  |<------ */
+    /*             |                                | */
+
+    /* There are three cases.  Note these are different from those
+       given in the LPC-10e phase 1 report. */
+
+    /* 1.  If there are no onsets in this range, then the voicing window
+       is centered in the pitch window.  If such a placement is not within
+       the window's placement range, then the window is placed in the left-most
+       portion of the placement range.  Its length is always MAXWIN. */
+
+    /* 2.  If the first onset is in 2F and there is sufficient room to place
+       the window immediately before this onset, then the window is placed
+       there, and its length is set to the maximum possible under these
+       constraints. */
+
+    /* "Critical Region Exception":  If there is another onset in 2F
+       such that a window can be placed between the two onsets, the
+       window is placed there (ie, as in case 3). */
+
+    /* 3.  Otherwise, the window is placed immediately after the onset.  The
+       window's length is the longest length that can fit in the range under these
+       constraints, except that the window may be shortened even further to avoid
+       overlapping other onsets in the placement range.  In any case, the window's
+       length is at least MINWIN. */
+
+    /* Note that the values of MINWIN and LFRAME must be chosen such
+       that case 2 = false implies case 3 = true.   This means that
+       MINWIN <= LFRAME/2.  If this were not the case, then a fourth case
+       would have to be added for when the window cannot fit either before
+       or after the onset. */
+
+    /* Note also that onsets which weren't in 2F last time may be in 1F this
+       time, due to the filter delays in computing onsets.  The result is that
+       occasionally a voicing window will overlap that onset.  The only way
+       to circumvent this problem is to add more delay in processing input
+       speech.  In the trade-off between delay and window-placement, window
+       placement lost. */
+
+    /* Compute the placement range */
+
+    /* Computing MAX */
+    i1 = vwin[af - 2][1] + 1;
+    i2 = (af - 2)*lframe + 1;
+    lrange = max(i1, i2);
+    hrange = af*lframe;
+    /* Compute OSPTR1, so the following code only looks at relevant onsets. */
+    for (osptr1 = *osptr - 1;  osptr1 >= 1;  osptr1--)
+    {
+        if (osbuf[osptr1 - 1] <= hrange)
+            break;
+    }
+    osptr1++;
+    /* Check for case 1 first (fast case) */
+    if (osptr1 <= 1  ||  osbuf[osptr1 - 2] < lrange)
+    {
+        /* Compute max */
+        i1 = vwin[af - 2][1] + 1;
+        vwin[af - 1][0] = max(i1, dvwinl);
+        vwin[af - 1][1] = vwin[af - 1][0] + maxwin - 1;
+        *obound = 0;
+    }
+    else
+    {
+        /* Search backward in OSBUF for first onset in range. */
+        /* This code relies on the above check being performed first. */
+        for (q = osptr1 - 1;  q >= 1;  q--)
+        {
+            if (osbuf[q - 1] < lrange)
+                break;
+        }
+        q++;
+        /* Check for case 2 (placement before onset): */
+        /* Check for critical region exception: */
+        crit = FALSE;
+        for (i = q + 1;  i < osptr1;  i++)
+        {
+            if (osbuf[i - 1] - osbuf[q - 1] >= minwin)
+            {
+                crit = TRUE;
+                break;
+            }
+        }
+        /* Compute max */
+        i1 = (af - 1)*lframe;
+        i2 = lrange + minwin - 1;
+        if (!crit  &&  osbuf[q - 1] > max(i1, i2))
+        {
+            vwin[af - 1][1] = osbuf[q - 1] - 1;
+            /* Compute max */
+            i2 = vwin[af - 1][1] - maxwin + 1;
+            vwin[af - 1][0] = max(lrange, i2);
+            *obound = 2;
+        }
+        else
+        {
+            /* Case 3 (placement after onset) */
+            vwin[af - 1][0] = osbuf[q - 1];
+            do
+            {
+                if (++q >= osptr1
+                    ||
+                    osbuf[q - 1] > vwin[af - 1][0] + maxwin)
+                {
+                    /* Compute min */
+                    i1 = vwin[af - 1][0] + maxwin - 1;
+                    vwin[af - 1][1] = min(i1, hrange);
+                    *obound = 1;
+                    return;
+                }
+            }
+            while (osbuf[q - 1] < vwin[af - 1][0] + minwin);
+            vwin[af - 1][1] = osbuf[q - 1] - 1;
+            *obound = 3;
+        }
+    }
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/lpc10_voicing.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/lpc10_voicing.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,487 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * lpc10_voicing.c - LPC10 low bit rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * This code is based on the U.S. Department of Defense reference
+ * implementation of the LPC-10 2400 bps Voice Coder. They do not
+ * exert copyright claims on their code, and it may be freely used.
+ *
+ * $Id: lpc10_voicing.c,v 1.7 2006/11/30 15:41:47 steveu Exp $
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <inttypes.h>
+#include <memory.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/dc_restore.h"
+#include "voipcodecs/lpc10.h"
+
+#include "lpc10_encdecs.h"
+
+static void vparms(int32_t vwin[],
+                    float *inbuf,
+                    float *lpbuf,
+                    const int32_t buflim[],
+                    int32_t half,
+                    float *dither,
+                    int32_t *mintau,
+                    int32_t *zc, 
+                    int32_t *lbe,
+                    int32_t *fbe,
+                    float *qs,
+                    float *rc1,
+                    float *ar_b,
+                    float *ar_f)
+{
+    int32_t inbuf_offset;
+    int32_t lpbuf_offset;
+    int32_t vlen;
+    int32_t stop;
+    int32_t i;
+    int32_t start;
+    float r1;
+    float r2;
+    float e_pre;
+    float ap_rms;
+    float e_0;
+    float oldsgn;
+    float lp_rms;
+    float e_b;
+    float e_f;
+    float r_b;
+    float r_f;
+    float e0ap;
+
+    /* Calculate zero crossings (ZC) and several energy and correlation */
+    /* measures on low band and full band speech.  Each measure is taken */
+    /* over either the first or the second half of the voicing window, */
+    /* depending on the variable HALF. */
+    lpbuf_offset = buflim[2];
+    lpbuf -= lpbuf_offset;
+    inbuf_offset = buflim[0];
+    inbuf -= inbuf_offset;
+
+    lp_rms = 0.0f;
+    ap_rms = 0.0f;
+    e_pre = 0.0f;
+    e0ap = 0.0f;
+    *rc1 = 0.0f;
+    e_0 = 0.0f;
+    e_b = 0.0f;
+    e_f = 0.0f;
+    r_f = 0.0f;
+    r_b = 0.0f;
+    *zc = 0;
+    vlen = vwin[1] - vwin[0] + 1;
+    start = vwin[0] + half*vlen/2 + 1;
+    stop = start + vlen/2 - 1;
+
+    /* I'll use the symbol HVL in the table below to represent the value */
+    /* VLEN/2.  Note that if VLEN is odd, then HVL should be rounded down, */
+    /* i.e., HVL = (VLEN-1)/2. */
+
+    /* HALF  START          STOP */
+
+    /* 1     VWIN(1)+1      VWIN(1)+HVL */
+    /* 2     VWIN(1)+HVL+1  VWIN(1)+2*HVL */
+    oldsgn = r_sign(1.0f, inbuf[start - 1] - *dither);
+    for (i = start;  i <= stop;  i++)
+    {
+        lp_rms += fabsf(lpbuf[i]);
+        ap_rms += fabsf(inbuf[i]);
+        e_pre += fabsf(inbuf[i] - inbuf[i - 1]);
+        r1 = inbuf[i];
+        e0ap += r1*r1;
+        *rc1 += inbuf[i]*inbuf[i - 1];
+        r1 = lpbuf[i];
+        e_0 += r1*r1;
+        r1 = lpbuf[i - *mintau];
+        e_b += r1*r1;
+        r1 = lpbuf[i + *mintau];
+        e_f += r1*r1;
+        r_f += lpbuf[i]*lpbuf[i + *mintau];
+        r_b += lpbuf[i]*lpbuf[i - *mintau];
+        r1 = inbuf[i] + *dither;
+        if (r_sign(1.0f, r1) != oldsgn)
+        {
+            ++(*zc);
+            oldsgn = -oldsgn;
+        }
+        *dither = -(*dither);
+    }
+    /* Normalized short-term autocovariance coefficient at unit sample delay */
+    *rc1 /= max(e0ap, 1.0f);
+    /* Ratio of the energy of the first difference signal (6 dB/oct preemphasis)*/
+    /* to the energy of the full band signal */
+    /* Computing MAX */
+    r1 = ap_rms*2.0f;
+    *qs = e_pre/max(r1, 1.0f);
+    /* aR_b is the product of the forward and reverse prediction gains, */
+    /* looking backward in time (the causal case). */
+    *ar_b = r_b/max(e_b, 1.0f)*(r_b/max(e_0, 1.0f));
+    /* aR_f is the same as aR_b, but looking forward in time (non causal case).*/
+    *ar_f = r_f/max(e_f, 1.0f)*(r_f/max(e_0, 1.0f));
+    /* Normalize ZC, LBE, and FBE to old fixed window length of 180. */
+    /* (The fraction 90/VLEN has a range of 0.58 to 1) */
+    r2 = (float) (*zc << 1);
+    *zc = lrintf(r2*(90.0f/vlen));
+    r1 = lp_rms/4*(90.0f/vlen);
+    *lbe = min(lrintf(r1), 32767);
+    r1 = ap_rms/4*(90.0f/vlen);
+    *fbe = min(lrintf(r1), 32767);
+}
+/*- End of function --------------------------------------------------------*/
+
+/* Voicing detection makes voicing decisions for each half */
+/* frame of input speech.  Tentative voicing decisions are made two frames*/
+/* in the future (2F) for each half frame.  These decisions are carried */
+/* through one frame in the future (1F) to the present (P) frame where */
+/* they are examined and smoothed, resulting in the final voicing */
+/* decisions for each half frame. */
+
+/* The voicing parameter (signal measurement) column vector (VALUE) */
+/* is based on a rectangular window of speech samples determined by the */
+/* window placement algorithm.  The voicing parameter vector contains the*/
+/* AMDF windowed maximum-to-minimum ratio, the zero crossing rate, energy*/
+/* measures, reflection coefficients, and prediction gains.  The voicing */
+/* window is placed to avoid contamination of the voicing parameter vector*/
+/* with speech onsets. */
+
+/* The input signal is then classified as unvoiced (including */
+/* silence) or voiced.  This decision is made by a linear discriminant */
+/* function consisting of a dot product of the voicing decision */
+/* coefficient (VDC) row vector with the measurement column vector */
+/* (VALUE).  The VDC vector is 2-dimensional, each row vector is optimized*/
+/* for a particular signal-to-noise ratio (SNR).  So, before the dot */
+/* product is performed, the SNR is estimated to select the appropriate */
+/* VDC vector. */
+
+/* The smoothing algorithm is a modified median smoother.  The */
+/* voicing discriminant function is used by the smoother to determine how*/
+/* strongly voiced or unvoiced a signal is.  The smoothing is further */
+/* modified if a speech onset and a voicing decision transition occur */
+/* within one half frame.  In this case, the voicing decision transition */
+/* is extended to the speech onset.  For transmission purposes, there are*/
+/* constraints on the duration and transition of voicing decisions.  The */
+/* smoother takes these constraints into account. */
+
+/* Finally, the energy estimates are updated along with the dither */
+/* threshold used to calculate the zero crossing rate (ZC). */
+
+void lpc10_voicing(lpc10_encode_state_t *s,
+                   int32_t vwin[],
+                   float *inbuf,
+                   float *lpbuf,
+                   const int32_t buflim[],
+                   int32_t half,
+                   float *minamd,
+                   float *maxamd, 
+                   int32_t *mintau,
+                   float ivrc[],
+                   int32_t obound[])
+{
+    static const float vdc[100] =
+    {
+        0.0f, 1714.0f, -110.0f, 334.0f, -4096.0f,  -654.0f, 3752.0f, 3769.0f, 0.0f,  1181.0f,
+        0.0f,  874.0f,  -97.0f, 300.0f, -4096.0f, -1021.0f, 2451.0f, 2527.0f, 0.0f,  -500.0f,
+        0.0f,  510.0f,  -70.0f, 250.0f, -4096.0f, -1270.0f, 2194.0f, 2491.0f, 0.0f, -1500.0f,
+        0.0f,  500.0f,  -10.0f, 200.0f, -4096.0f, -1300.0f,  2.0e3f,  2.0e3f, 0.0f,  -2.0e3f,
+        0.0f,  500.0f,    0.0f,   0.0f, -4096.0f, -1300.0f,  2.0e3f,  2.0e3f, 0.0f, -2500.0f,
+        0.0f,    0.0f,    0.0f,   0.0f,     0.0f,     0.0f,    0.0f,    0.0f, 0.0f,     0.0f,
+        0.0f,    0.0f,    0.0f,   0.0f,     0.0f,     0.0f,    0.0f,    0.0f, 0.0f,     0.0f,
+        0.0f,    0.0f,    0.0f,   0.0f,     0.0f,     0.0f,    0.0f,    0.0f, 0.0f,     0.0f,
+        0.0f,    0.0f,    0.0f,   0.0f,     0.0f,     0.0f,    0.0f,    0.0f, 0.0f,     0.0f,
+        0.0f,    0.0f,    0.0f,   0.0f,     0.0f,     0.0f,    0.0f,    0.0f, 0.0f,     0.0f
+    };
+    static const int nvdcl = 5;
+    static const float vdcl[10] =
+    {
+        600.0f, 450.0f, 300.0f, 200.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f, 0.0f
+    };
+
+    int32_t inbuf_offset;
+    int32_t lpbuf_offset;
+    int32_t i1;
+    float r1;
+    float r2;
+    float ar_b;
+    float ar_f;
+    int32_t snrl;
+    int32_t i;
+    float value[9];
+    int32_t zc;
+    int ot;
+    float qs;
+    int32_t vstate;
+    float rc1;
+    int32_t fbe;
+    int32_t lbe;
+    float snr2;
+
+    inbuf_offset = 0;
+    lpbuf_offset = 0;
+    if (inbuf)
+    {
+        inbuf_offset = buflim[0];
+        inbuf -= inbuf_offset;
+    }
+    if (lpbuf)
+    {
+        lpbuf_offset = buflim[2];
+        lpbuf -= lpbuf_offset;
+    }
+
+    /* Voicing Decision Parameter vector (* denotes zero coefficient): */
+
+    /*     * MAXMIN */
+    /*       LBE/LBVE */
+    /*       ZC */
+    /*       RC1 */
+    /*       QS */
+    /*       IVRC2 */
+    /*       aR_B */
+    /*       aR_F */
+    /*     * LOG(LBE/LBVE) */
+    /* Define 2-D voicing decision coefficient vector according to the voicing */
+    /* parameter order above.  Each row (VDC vector) is optimized for a specific */
+    /*   SNR.  The last element of the vector is the constant. */
+    /*              E    ZC    RC1    Qs   IVRC2  aRb   aRf        c */
+
+    /* The VOICE array contains the result of the linear discriminant function*/
+    /* (analog values).  The VOIBUF array contains the hard-limited binary */
+    /* voicing decisions.  The VOICE and VOIBUF arrays, according to FORTRAN */
+    /* memory allocation, are addressed as: */
+
+    /*        (half-frame number, future-frame number) */
+
+    /*        |   Past    |  Present  |  Future1  |  Future2  | */
+    /*        | 1,0 | 2,0 | 1,1 | 2,1 | 1,2 | 2,2 | 1,3 | 2,3 |  --->  time */
+
+    /* Update linear discriminant function history each frame: */
+    if (half == 0)
+    {
+        s->voice[0][0] = s->voice[1][0];
+        s->voice[0][1] = s->voice[1][1];
+        s->voice[1][0] = s->voice[2][0];
+        s->voice[1][1] = s->voice[2][1];
+        s->maxmin = *maxamd / max(*minamd, 1.0f);
+    }
+    /* Calculate voicing parameters twice per frame */
+    vparms(vwin,
+           &inbuf[inbuf_offset],
+           &lpbuf[lpbuf_offset],
+           buflim, 
+           half,
+           &s->dither,
+           mintau,
+           &zc,
+           &lbe,
+           &fbe,
+           &qs,
+           &rc1,
+           &ar_b,
+           &ar_f);
+    /* Estimate signal-to-noise ratio to select the appropriate VDC vector. */
+    /* The SNR is estimated as the running average of the ratio of the */
+    /* running average full-band voiced energy to the running average */
+    /* full-band unvoiced energy. SNR filter has gain of 63. */
+    r1 = (s->snr + s->fbve/(float) max(s->fbue, 1))*63/64.0f;
+    s->snr = (float) lrintf(r1);
+    snr2 = s->snr*s->fbue/max(s->lbue, 1);
+    /* Quantize SNR to SNRL according to VDCL thresholds. */
+    i1 = nvdcl - 1;
+    for (snrl = 0;  snrl < i1;  snrl++)
+    {
+        if (snr2 > vdcl[snrl])
+            break;
+    }
+    /* (Note:  SNRL = NVDCL here) */
+    /* Linear discriminant voicing parameters: */
+    value[0] = s->maxmin;
+    value[1] = (float) lbe/max(s->lbve, 1);
+    value[2] = (float) zc;
+    value[3] = rc1;
+    value[4] = qs;
+    value[5] = ivrc[1];
+    value[6] = ar_b;
+    value[7] = ar_f;
+    /* Evaluation of linear discriminant function: */
+    s->voice[2][half] = vdc[snrl*10 + 9];
+    for (i = 0;  i < 8;  i++)
+        s->voice[2][half] += vdc[snrl*10 + i]*value[i];
+    /* Classify as voiced if discriminant > 0, otherwise unvoiced */
+    /* Voicing decision for current half-frame:  1 = Voiced; 0 = Unvoiced */
+    s->voibuf[3][half] = (s->voice[2][half] > 0.0f)  ?  1  :  0;
+    /* Skip voicing decision smoothing in first half-frame: */
+    /* Give a value to VSTATE, so that trace statements below will print */
+    /* a consistent value from one call to the next when HALF .EQ. 1. */
+    /* The value of VSTATE is not used for any other purpose when this is */
+    /* true. */
+    vstate = -1;
+    if (half != 0)
+    {
+        /* Voicing decision smoothing rules (override of linear combination): */
+
+        /*     Unvoiced half-frames:  At least two in a row. */
+        /*     -------------------- */
+
+        /*     Voiced half-frames:    At least two in a row in one frame. */
+        /*     -------------------    Otherwise at least three in a row. */
+        /*                    (Due to the way transition frames are encoded) */
+
+        /* In many cases, the discriminant function determines how to smooth. */
+        /* In the following chart, the decisions marked with a * may be overridden. */
+
+        /* Voicing override of transitions at onsets: */
+        /* If a V/UV or UV/V voicing decision transition occurs within one-half */
+        /* frame of an onset bounding a voicing window, then the transition is */
+        /* moved to occur at the onset. */
+
+        /*     P    1F */
+        /*     -----    ----- */
+        /*     0   0   0   0 */
+        /*     0   0   0*  1    (If there is an onset there) */
+        /*     0   0   1*  0*    (Based on 2F and discriminant distance) */
+        /*     0   0   1   1 */
+        /*     0   1*  0   0    (Always) */
+        /*     0   1*  0*  1    (Based on discriminant distance) */
+        /*     0*  1   1   0*    (Based on past, 2F, and discriminant distance) */
+        /*     0   1*  1   1    (If there is an onset there) */
+        /*     1   0*  0   0    (If there is an onset there) */
+        /*     1   0   0   1 */
+        /*     1   0*  1*  0    (Based on discriminant distance) */
+        /*     1   0*  1   1    (Always) */
+        /*     1   1   0   0 */
+        /*     1   1   0*  1*    (Based on 2F and discriminant distance) */
+        /*     1   1   1*  0    (If there is an onset there) */
+        /*     1   1   1   1 */
+
+        /* Determine if there is an onset transition between P and 1F. */
+        /* OT (Onset Transition) is true if there is an onset between */
+        /* P and 1F but not after 1F. */
+        ot = ((obound[0] & 2) != 0  ||  obound[1] == 1)  &&  (obound[2] & 1) == 0;
+        /* Multi-way dispatch on voicing decision history: */
+        vstate = (s->voibuf[1][0] << 3) + (s->voibuf[1][1] << 2) + (s->voibuf[2][0] << 1) + s->voibuf[2][1];
+        switch (vstate + 1)
+        {
+        case 2:
+            if (ot  &&  s->voibuf[3][0] == 1)
+                s->voibuf[2][0] = 1;
+            break;
+        case 3:
+            if (s->voibuf[3][0] == 0  ||  s->voice[1][0] < -s->voice[1][1])
+                s->voibuf[2][0] = 0;
+            else
+                s->voibuf[2][1] = 1;
+            break;
+        case 5:
+            s->voibuf[1][1] = 0;
+            break;
+        case 6:
+            if (s->voice[0][1] < -s->voice[1][0])
+                s->voibuf[1][1] = 0;
+            else
+                s->voibuf[2][0] = 1;
+            break;
+        case 7:
+            if (s->voibuf[0][0] == 1  ||  s->voibuf[3][0] == 1  ||  s->voice[1][1] > s->voice[0][0])
+                s->voibuf[2][1] = 1;
+            else
+                s->voibuf[1][0] = 1;
+            break;
+        case 8:
+            if (ot)
+                s->voibuf[1][1] = 0;
+            break;
+        case 9:
+            if (ot)
+                s->voibuf[1][1] = 1;
+            break;
+        case 11:
+            if (s->voice[1][9] < -s->voice[0][1])
+                s->voibuf[2][0] = 0;
+            else
+                s->voibuf[1][1] = 1;
+            break;
+        case 12:
+            s->voibuf[1][1] = 1;
+            break;
+        case 14:
+            if (s->voibuf[3][0] == 0  &&  s->voice[1][1] < -s->voice[1][0])
+                s->voibuf[2][1] = 0;
+            else
+                s->voibuf[2][0] = 1;
+            break;
+        case 15:
+            if (ot  &&  s->voibuf[3][0] == 0)
+                s->voibuf[2][0] = 0;
+            break;
+        }
+    }
+    /* During unvoiced half-frames, update the low band and full band unvoiced*/
+    /* energy estimates (LBUE and FBUE) and also the zero crossing */
+    /* threshold (DITHER).  (The input to the unvoiced energy filters is */
+    /* restricted to be less than 10dB above the previous inputs of the */
+    /* filters.) */
+    /* During voiced half-frames, update the low-pass (LBVE) and all-pass */
+    /* (FBVE) voiced energy estimates. */
+    if (s->voibuf[3][half] == 0)
+    {
+        r1 = (s->sfbue*63 + (min(fbe, s->ofbue*3) << 3))/64.0f;
+        s->sfbue = lrintf(r1);
+        s->fbue = s->sfbue/8;
+        s->ofbue = fbe;
+        r1 = (s->slbue*63 + (min(lbe, s->olbue*3) << 3))/64.0f;
+        s->slbue = lrintf(r1);
+        s->lbue = s->slbue/8;
+        s->olbue = lbe;
+    }
+    else
+    {
+        s->lbve = lrintf((s->lbve*63 + lbe)/64.0f);
+        s->fbve = lrintf((s->fbve*63 + fbe)/64.0f);
+    }
+    /* Set dither threshold to yield proper zero crossing rates in the */
+    /* presence of low frequency noise and low level signal input. */
+    /* NOTE: The divisor is a function of REF, the expected energies. */
+    /* Computing MIN */
+    /* Computing MAX */
+    r2 = sqrtf((float) (s->lbue*s->lbve))*64/3000;
+    r1 = max(r2, 1.0f);
+    s->dither = min(r1, 20.0f);
+    /* Voicing decisions are returned in VOIBUF. */
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/msvc/gettimeofday.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/msvc/gettimeofday.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,14 @@
+#ifdef _MSC_VER
+#pragma warning(disable:4100)
+#endif
+
+#include "windows.h"
+
+void gettimeofday(struct timeval *tv, void *tz)
+{
+    long int l = GetTickCount();
+ 
+    tv->tv_sec = l / 1000;
+    tv->tv_usec = (l % 1000) * 1000;
+    return;
+} 

Added: freeswitch/trunk/libs/voipcodecs/src/msvc/inttypes.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/msvc/inttypes.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,87 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * inttypes.h - a fudge for MSVC, which lacks this header
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Michael Jerris
+ *
+ *
+ * This file is released in the public domain.
+ *
+ */
+
+#if !defined(_INTTYPES_H_)
+#define _INTTYPES_H_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#ifdef _MSC_VER
+#if (_MSC_VER >= 1400) // VC8+
+#ifndef _CRT_SECURE_NO_DEPRECATE
+#define _CRT_SECURE_NO_DEPRECATE
+#endif
+#ifndef _CRT_NONSTDC_NO_DEPRECATE
+#define _CRT_NONSTDC_NO_DEPRECATE
+#endif
+#endif // VC8+
+#include <windows.h>
+typedef unsigned __int8		uint8_t;
+typedef unsigned __int16	uint16_t;
+typedef unsigned __int32	uint32_t;
+typedef unsigned __int64    uint64_t;
+typedef __int8		int8_t;
+typedef __int16		int16_t;
+typedef __int32		int32_t;
+typedef __int64		int64_t;
+#define inline __inline
+#define __inline__ __inline
+#define  INT16_MAX   0x7fff 
+#define  INT16_MIN   (-INT16_MAX - 1) 
+#define _MMX_H_
+
+/* disable the following warnings 
+ * C4100: The formal parameter is not referenced in the body of the function. The unreferenced parameter is ignored. 
+ * C4200: Non standard extension C zero sized array
+ * C4706: assignment within conditional expression
+ * C4244: conversion from 'type1' to 'type2', possible loss of data
+ * C4295: array is too small to include a terminating null character
+ * C4125: decimal digit terminates octal escape sequence
+ */
+#pragma warning(disable:4100 4200 4706 4295 4125)
+
+#pragma comment(lib, "ws2_32.lib")
+
+#define strncasecmp _strnicmp
+#define strcasecmp _stricmp
+#define snprintf _snprintf
+
+#if !defined(INFINITY)
+#define INFINITY 0x7fffffff
+#endif
+#endif
+
+#define PACKAGE "voipcodecs"
+#define VERSION "0.0.1andabit"
+
+#define INT32_MAX	(2147483647)
+#define INT32_MIN	(-2147483647 - 1)
+
+#define PRId8 "d"
+#define PRId16 "d"
+#define PRId32 "ld"
+#define PRId64 "lld"
+
+#define PRIu8 "u"
+#define PRIu16 "u"
+#define PRIu32 "lu"
+#define PRIu64 "llu"
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

Added: freeswitch/trunk/libs/voipcodecs/src/msvc/msvcproj.foot
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/msvc/msvcproj.foot	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,7 @@
+
+# Begin Group "Resource Files"
+
+# PROP Default_Filter "ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe"
+# End Group
+# End Target
+# End Project

Added: freeswitch/trunk/libs/voipcodecs/src/msvc/msvcproj.head
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/msvc/msvcproj.head	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,92 @@
+# Microsoft Developer Studio Project File - Name="voipcodecs" - Package Owner=<4>
+# Microsoft Developer Studio Generated Build File, Format Version 6.00
+# ** DO NOT EDIT **
+
+# TARGTYPE "Win32 (x86) Dynamic-Link Library" 0x0102
+
+CFG=voipcodecs - Win32 Debug
+!MESSAGE This is not a valid makefile. To build this project using NMAKE,
+!MESSAGE use the Export Makefile command and run
+!MESSAGE 
+!MESSAGE NMAKE /f "voipcodecs.mak".
+!MESSAGE 
+!MESSAGE You can specify a configuration when running NMAKE
+!MESSAGE by defining the macro CFG on the command line. For example:
+!MESSAGE 
+!MESSAGE NMAKE /f "voipcodecs.mak" CFG="voipcodecs - Win32 Debug"
+!MESSAGE 
+!MESSAGE Possible choices for configuration are:
+!MESSAGE 
+!MESSAGE "voipcodecs - Win32 Release" (based on "Win32 (x86) Dynamic-Link Library")
+!MESSAGE "voipcodecs - Win32 Debug" (based on "Win32 (x86) Dynamic-Link Library")
+!MESSAGE 
+
+# Begin Project
+# PROP AllowPerConfigDependencies 0
+# PROP Scc_ProjName ""
+# PROP Scc_LocalPath ""
+CPP=cl.exe
+MTL=midl.exe
+RSC=rc.exe
+
+!IF  "$(CFG)" == "voipcodecs - Win32 Release"
+
+# PROP BASE Use_MFC 0
+# PROP BASE Use_Debug_Libraries 0
+# PROP BASE Output_Dir "Release"
+# PROP BASE Intermediate_Dir "Release"
+# PROP BASE Target_Dir ""
+# PROP Use_MFC 0
+# PROP Use_Debug_Libraries 0
+# PROP Output_Dir "Release"
+# PROP Intermediate_Dir "Release"
+# PROP Ignore_Export_Lib 0
+# PROP Target_Dir ""
+# ADD BASE CPP /nologo /MT /W3 /GX /O2 /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D HAVE_TGMATH_H /YX /FD /c
+# ADD CPP /nologo /MT /W3 /GX /Zi /O2 /I "." /I "..\include" /D "WIN32" /D "NDEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D HAVE_TGMATH_H /D "_WINDLL" /FR /FD /c
+# SUBTRACT CPP /YX
+# ADD BASE MTL /nologo /D "NDEBUG" /mktyplib203 /win32
+# ADD MTL /nologo /D "NDEBUG" /mktyplib203 /win32
+# ADD BASE RSC /l 0x409 /d "NDEBUG"
+# ADD RSC /l 0x409 /d "NDEBUG"
+BSC32=bscmake.exe
+# ADD BASE BSC32 /nologo
+# ADD BSC32 /nologo
+LINK32=link.exe
+# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /dll /machine:I386
+# ADD LINK32 kernel32.lib ws2_32.lib winmm.lib /nologo /dll /map /debug /machine:I386 /out:"Release/libvoipcodecs.dll"
+
+!ELSEIF  "$(CFG)" == "voipcodecs - Win32 Debug"
+
+# PROP BASE Use_MFC 0
+# PROP BASE Use_Debug_Libraries 1
+# PROP BASE Output_Dir "Debug"
+# PROP BASE Intermediate_Dir "Debug"
+# PROP BASE Target_Dir ""
+# PROP Use_MFC 0
+# PROP Use_Debug_Libraries 1
+# PROP Output_Dir "Debug"
+# PROP Intermediate_Dir "Debug"
+# PROP Ignore_Export_Lib 0
+# PROP Target_Dir ""
+# ADD BASE CPP /nologo /MTd /W3 /Gm /GX /ZI /Od /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D HAVE_TGMATH_H /YX /FD /GZ /c
+# ADD CPP /nologo /MTd /W3 /Gm /GX /Zi /Od /I "." /I "..\include" /D "WIN32" /D "_DEBUG" /D "_WINDOWS" /D "_MBCS" /D "_USRDLL" /D HAVE_TGMATH_H /FR /FD /GZ /c
+# SUBTRACT CPP /WX /YX
+# ADD BASE MTL /nologo /D "_DEBUG" /mktyplib203 /win32
+# ADD MTL /nologo /D "_DEBUG" /mktyplib203 /win32
+# ADD BASE RSC /l 0x409 /d "_DEBUG"
+# ADD RSC /l 0x409 /d "_DEBUG"
+BSC32=bscmake.exe
+# ADD BASE BSC32 /nologo
+# ADD BSC32 /nologo
+LINK32=link.exe
+# ADD BASE LINK32 kernel32.lib user32.lib gdi32.lib winspool.lib comdlg32.lib advapi32.lib shell32.lib ole32.lib oleaut32.lib uuid.lib odbc32.lib odbccp32.lib /nologo /dll /debug /machine:I386 /pdbtype:sept
+# ADD LINK32 kernel32.lib ws2_32.lib winmm.lib /nologo /dll /incremental:no /map /debug /machine:I386 /out:"Debug/libvoipcodecs.dll" /pdbtype:sept
+# SUBTRACT LINK32 /nodefaultlib
+
+!ENDIF 
+
+# Begin Target
+
+# Name "voipcodecs - Win32 Release"
+# Name "voipcodecs - Win32 Debug"

Added: freeswitch/trunk/libs/voipcodecs/src/msvc/sys/time.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/msvc/sys/time.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1 @@
+extern void gettimeofday(struct timeval *tv, void *tz);

Added: freeswitch/trunk/libs/voipcodecs/src/msvc/tgmath.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/msvc/tgmath.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,84 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * tgmath.h - a fudge for MSVC, which lacks this header
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Michael Jerris
+ *
+ *
+ * This file is released in the public domain.
+ *
+ */
+
+#if !defined(_TGMATH_H_)
+#define _TGMATH_H_
+
+#include <math.h>
+
+#if !defined(M_PI)
+/* C99 systems may not define M_PI */
+#define M_PI 3.14159265358979323846264338327
+#endif
+
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+/* A kindofa rint() for VC++ (only kindofa, because rint should be type generic,
+   and this one is purely float to int */
+static inline long int lrintf(float a)
+{
+    long int i;
+    
+    __asm
+    {
+        fld   a
+        fistp i
+    }
+    return i;
+}
+
+static inline long int lrint(double a)
+{
+    long int i;
+    
+    __asm
+    {
+        fld   a
+        fistp i
+    }
+    return i;
+}
+
+static inline int rintf(float a)
+{
+    int i;
+    
+    __asm
+    {
+        fld   a
+        fistp i
+    }
+    return i;
+}
+
+static inline int rint(double a)
+{
+    int i;
+    
+    __asm
+    {
+        fld   a
+        fistp i
+    }
+    return i;
+}
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

Added: freeswitch/trunk/libs/voipcodecs/src/msvc/unistd.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/msvc/unistd.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,31 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * inttypes.h - a fudge for MSVC, which lacks this header
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Michael Jerris
+ *
+ *
+ * This file is released in the public domain.
+ *
+ */
+
+#if !defined(_INTTYPES_H_)
+#define _INTTYPES_H_
+
+#ifdef __cplusplus
+extern "C" {
+#endif
+
+#define open _open
+#define write _write
+
+extern int gethostname (char *name, size_t len);
+
+#ifdef __cplusplus
+}
+#endif
+
+#endif

Added: freeswitch/trunk/libs/voipcodecs/src/msvc/vc8proj.foot
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/msvc/vc8proj.foot	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,11 @@
+		</Filter>
+		<Filter
+			Name="Resource Files"
+			Filter="rc;ico;cur;bmp;dlg;rc2;rct;bin;rgs;gif;jpg;jpeg;jpe;resx;tiff;tif;png;wav"
+			UniqueIdentifier="{67DA6AB6-F800-4c08-8B7A-83BB121AAD01}"
+			>
+		</Filter>
+	</Files>
+	<Globals>
+	</Globals>
+</VisualStudioProject>

Added: freeswitch/trunk/libs/voipcodecs/src/msvc/vc8proj.head
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/msvc/vc8proj.head	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,180 @@
+<?xml version="1.0" encoding="Windows-1252"?>
+<VisualStudioProject
+	ProjectType="Visual C++"
+	Version="8.00"
+	Name="libvoipcodecs"
+	ProjectGUID="{CF70F278-3364-4395-A2E1-23501C9B8AD2}"
+	RootNamespace="libvoipcodecs"
+	Keyword="Win32Proj"
+	>
+	<Platforms>
+		<Platform
+			Name="Win32"
+		/>
+	</Platforms>
+	<ToolFiles>
+	</ToolFiles>
+	<Configurations>
+		<Configuration
+			Name="Debug|Win32"
+			OutputDirectory="$(SolutionDir)$(ConfigurationName)"
+			IntermediateDirectory="$(ConfigurationName)"
+			ConfigurationType="2"
+			CharacterSet="1"
+			>
+			<Tool
+				Name="VCPreBuildEventTool"
+			/>
+			<Tool
+				Name="VCCustomBuildTool"
+			/>
+			<Tool
+				Name="VCXMLDataGeneratorTool"
+			/>
+			<Tool
+				Name="VCWebServiceProxyGeneratorTool"
+			/>
+			<Tool
+				Name="VCMIDLTool"
+			/>
+			<Tool
+				Name="VCCLCompilerTool"
+				Optimization="0"
+				AdditionalIncludeDirectories="src; src\voipcodecs; src\msvc"
+				PreprocessorDefinitions="WIN32;_DEBUG;_WINDOWS;_USRDLL;LIBSPANDSP_EXPORTS;_CRT_SECURE_NO_DEPRECATE;_CRT_NONSTDC_NO_DEPRECATE;HAVE_TGMATH_H"
+				MinimalRebuild="true"
+				BasicRuntimeChecks="3"
+				RuntimeLibrary="3"
+				UsePrecompiledHeader="0"
+				WarningLevel="4"
+				Detect64BitPortabilityProblems="true"
+				DebugInformationFormat="4"
+			/>
+			<Tool
+				Name="VCManagedResourceCompilerTool"
+			/>
+			<Tool
+				Name="VCResourceCompilerTool"
+			/>
+			<Tool
+				Name="VCPreLinkEventTool"
+			/>
+			<Tool
+				Name="VCLinkerTool"
+				LinkIncremental="2"
+				ModuleDefinitionFile="src/msvc/voipcodecs.def"
+				GenerateDebugInformation="true"
+				SubSystem="2"
+				TargetMachine="1"
+			/>
+			<Tool
+				Name="VCALinkTool"
+			/>
+			<Tool
+				Name="VCManifestTool"
+			/>
+			<Tool
+				Name="VCXDCMakeTool"
+			/>
+			<Tool
+				Name="VCBscMakeTool"
+			/>
+			<Tool
+				Name="VCFxCopTool"
+			/>
+			<Tool
+				Name="VCAppVerifierTool"
+			/>
+			<Tool
+				Name="VCWebDeploymentTool"
+			/>
+			<Tool
+				Name="VCPostBuildEventTool"
+			/>
+		</Configuration>
+		<Configuration
+			Name="Release|Win32"
+			OutputDirectory="$(SolutionDir)$(ConfigurationName)"
+			IntermediateDirectory="$(ConfigurationName)"
+			ConfigurationType="2"
+			CharacterSet="1"
+			WholeProgramOptimization="1"
+			>
+			<Tool
+				Name="VCPreBuildEventTool"
+			/>
+			<Tool
+				Name="VCCustomBuildTool"
+			/>
+			<Tool
+				Name="VCXMLDataGeneratorTool"
+			/>
+			<Tool
+				Name="VCWebServiceProxyGeneratorTool"
+			/>
+			<Tool
+				Name="VCMIDLTool"
+			/>
+			<Tool
+				Name="VCCLCompilerTool"
+				AdditionalIncludeDirectories="src; src\voipcodecs; src\msvc"
+				PreprocessorDefinitions="WIN32;NDEBUG;_WINDOWS;_USRDLL;LIBSPANDSP_EXPORTS;_CRT_SECURE_NO_DEPRECATE;_CRT_NONSTDC_NO_DEPRECATE;HAVE_TGMATH_H"
+				RuntimeLibrary="2"
+				UsePrecompiledHeader="0"
+				WarningLevel="4"
+				Detect64BitPortabilityProblems="true"
+				DebugInformationFormat="3"
+			/>
+			<Tool
+				Name="VCManagedResourceCompilerTool"
+			/>
+			<Tool
+				Name="VCResourceCompilerTool"
+			/>
+			<Tool
+				Name="VCPreLinkEventTool"
+			/>
+			<Tool
+				Name="VCLinkerTool"
+				LinkIncremental="1"
+				ModuleDefinitionFile="src/msvc/voipcodecs.def"
+				GenerateDebugInformation="true"
+				SubSystem="2"
+				OptimizeReferences="2"
+				EnableCOMDATFolding="2"
+				TargetMachine="1"
+			/>
+			<Tool
+				Name="VCALinkTool"
+			/>
+			<Tool
+				Name="VCManifestTool"
+			/>
+			<Tool
+				Name="VCXDCMakeTool"
+			/>
+			<Tool
+				Name="VCBscMakeTool"
+			/>
+			<Tool
+				Name="VCFxCopTool"
+			/>
+			<Tool
+				Name="VCAppVerifierTool"
+			/>
+			<Tool
+				Name="VCWebDeploymentTool"
+			/>
+			<Tool
+				Name="VCPostBuildEventTool"
+			/>
+		</Configuration>
+	</Configurations>
+	<References>
+	</References>
+	<Files>
+		<Filter
+			Name="Source Files"
+			Filter="cpp;c;cc;cxx;def;odl;idl;hpj;bat;asm;asmx"
+			UniqueIdentifier="{4FC737F1-C7A5-4376-A066-2A32D752A2FF}"
+			>

Added: freeswitch/trunk/libs/voipcodecs/src/msvc/voipcodecs.def
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/msvc/voipcodecs.def	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,304 @@
+EXPORTS
+adsi_rx_init
+adsi_rx
+adsi_tx_init
+adsi_send_alert_tone
+adsi_put_message
+adsi_next_field
+adsi_add_field
+async_tx_init
+async_rx_init
+awgn
+bert_init
+bert_get_bit
+bert_put_bit
+bert_set_report
+bert_result
+filter_create
+filter_delete
+filter_step
+cfilter_create
+cfilter_delete
+cfilter_step
+dds_phase_rate
+dds_frequency
+dds_scaling_dbm0
+dds_scaling_dbov
+dds_lookup
+dds_offset
+dds
+dds_mod
+dds_complex
+dds_complex_mod
+dds_phase_ratef
+dds_frequencyf
+dds_scaling_dbm0f
+dds_scaling_dbovf
+ddsf
+dds_modf
+dds_complexf
+dds_complex_modf
+echo_can_create
+echo_can_free
+echo_can_flush
+echo_can_adaption_mode
+echo_can_update
+fax_rx
+fax_tx
+fax_set_flush_handler
+fax_init
+fax_release
+fsk_tx_init
+fsk_tx_power
+fsk_tx_set_get_bit
+fsk_tx
+fsk_rx_signal_power
+fsk_rx_signal_cutoff
+fsk_rx_init
+fsk_rx
+fsk_rx_set_put_bit
+g722_encode_init
+g722_encode_release
+g722_encode
+g722_decode_init
+g722_decode_release
+g722_decode
+g726_init
+g726_release
+crc_itu32_calc
+crc_itu32_append
+crc_itu32_check
+crc_itu16_calc
+crc_itu16_append
+crc_itu16_check
+hdlc_rx_init
+hdlc_rx_get_stats
+hdlc_tx_init
+hdlc_tx_frame
+hdlc_tx_preamble
+ima_adpcm_init
+ima_adpcm_release
+span_log_test
+span_log
+span_log_buf
+span_log_init
+span_log_set_protocol
+span_set_message_handler
+span_set_error_handler
+modem_echo_can_create
+modem_echo_can_free
+modem_echo_can_flush
+modem_echo_can_adaption_mode
+modem_echo_can_update
+noise
+oki_adpcm_init
+oki_adpcm_release
+playout_put
+playout_get
+playout_get_unconditional
+playout_current_length
+playout_next_due
+playout_new
+playout_free
+playout_restart
+plc_rx
+plc_fillin
+plc_init
+plc_release
+power_meter_init
+power_meter_damping
+power_meter_update
+power_meter_dbm0
+power_meter_dbov
+power_meter_level_dbm0
+power_meter_level_dbov
+queue_empty
+queue_free_space
+queue_contents
+queue_flush
+queue_view
+queue_read
+queue_write
+queue_test_msg
+queue_read_msg
+queue_write_msg
+queue_create
+queue_delete
+sig_tone_init
+sig_tone_rx
+sig_tone_tx
+super_tone_rx_make_descriptor
+super_tone_rx_add_tone
+super_tone_rx_add_element
+super_tone_rx_init
+super_tone_rx_free
+super_tone_rx_segment_callback
+super_tone_rx
+super_tone_tx_make_step
+super_tone_tx_free
+super_tone_tx_init
+super_tone_tx
+t30_init
+t30_release
+t30_restart
+t30_create
+t30_free
+t30_frametype
+t30_decode_dis_dtc_dcs
+t30_completion_code_to_str
+t30_set_header_info
+t30_set_local_ident
+t30_get_sub_address
+t30_get_header_info
+t30_get_local_ident
+t30_get_far_ident
+t30_get_transfer_statistics
+t30_set_phase_b_handler
+t30_set_phase_d_handler
+t30_set_phase_e_handler
+t30_set_document_handler
+t30_set_rx_file
+t30_set_tx_file
+t30_local_interrupt_request
+t30_send_complete
+t30_hdlc_accept
+t30_timer_update
+t31_call_event
+t31_at_rx
+t31_rx
+t31_tx
+t31_init
+t31_release
+t35_decode
+t38_indicator
+t38_data_type
+t38_field_type
+t38_terminal_init
+t38_gateway_init
+t4_rx_create
+t4_rx_init
+t4_rx_start_page
+t4_rx_end_page
+t4_rx_delete
+t4_rx_end
+t4_rx_set_rx_encoding
+t4_rx_set_sub_address
+t4_rx_set_far_ident
+t4_rx_set_vendor
+t4_rx_set_model
+t4_tx_create
+t4_tx_init
+t4_tx_start_page
+t4_tx_restart_page
+t4_tx_end_page
+t4_tx_delete
+t4_tx_end
+t4_tx_set_tx_encoding
+t4_tx_set_min_row_bits
+t4_tx_set_local_ident
+t4_tx_set_header_info
+t4_get_transfer_statistics
+t4_encoding_to_str
+time_scale_init
+time_scale_rate
+time_scale
+make_goertzel_descriptor
+goertzel_init
+goertzel_update
+goertzel_result
+dtmf_rx_init
+dtmf_rx_set_realtime_callback
+dtmf_rx_parms
+dtmf_rx
+make_tone_descriptor
+make_tone_gen_descriptor
+tone_gen_init
+tone_gen
+dtmf_tx_init
+dtmf_tx
+v17_rx_init
+v17_rx_restart
+v17_rx_release
+v17_rx_set_put_bit
+v17_rx
+v17_rx_equalizer_state
+v17_rx_carrier_frequency
+v17_rx_symbol_timing_correction
+v17_rx_signal_power
+v17_rx_signal_cutoff
+v17_rx_set_qam_report_handler
+v17_tx_power
+v17_tx_init
+v17_tx_restart
+v17_tx_release
+v17_tx_set_get_bit
+v17_tx
+v22bis_rx_restart
+v22bis_rx
+v22bis_rx_equalizer_state
+v22bis_rx_carrier_frequency
+v22bis_rx_symbol_timing_correction
+v22bis_rx_signal_power
+v22bis_rx_set_qam_report_handler
+v22bis_tx
+v22bis_tx_power
+v22bis_restart
+v22bis_init
+v22bis_set_get_bit
+v22bis_set_put_bit
+v27ter_rx_init
+v27ter_rx_restart
+v27ter_rx_release
+v27ter_rx_set_put_bit
+v27ter_rx
+v27ter_rx_equalizer_state
+v27ter_rx_carrier_frequency
+v27ter_rx_symbol_timing_correction
+v27ter_rx_signal_power
+v27ter_rx_signal_cutoff
+v27ter_rx_set_qam_report_handler
+v27ter_tx_power
+v27ter_tx_init
+v27ter_tx_restart
+v27ter_tx_release
+v27ter_tx_set_get_bit
+v27ter_tx
+v29_rx_init
+v29_rx_restart
+v29_rx_release
+v29_rx_set_put_bit
+v29_rx
+v29_rx_equalizer_state
+v29_rx_carrier_frequency
+v29_rx_symbol_timing_correction
+v29_rx_signal_power
+v29_rx_signal_cutoff
+v29_rx_set_qam_report_handler
+v29_tx_power
+v29_tx_init
+v29_tx_restart
+v29_tx_release
+v29_tx_set_get_bit
+v29_tx
+lapm_dump
+lapm_receive
+lapm_tx
+lapm_tx_iframe
+v42_set_status_callback
+v42_rx_bit
+v42_tx_bit
+v42_init
+v42_restart
+v42_release
+v42bis_compress
+v42bis_compress_flush
+v42bis_decompress
+v42bis_decompress_flush
+v42bis_init
+v42bis_release
+v8_init
+v8_release
+v8_tx
+v8_rx
+v8_log_supported_modulations
+

Added: freeswitch/trunk/libs/voipcodecs/src/oki_adpcm.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/oki_adpcm.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,376 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * oki_adpcm.c - Conversion routines between linear 16 bit PCM data and
+ *               OKI (Dialogic) ADPCM format. Supports with the 32kbps
+ *               and 24kbps variants used by Dialogic.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2001, 2004 Steve Underwood
+ *
+ * The actual OKI ADPCM encode and decode method is derived from freely
+ * available code, whose exact origins seem uncertain.
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: oki_adpcm.c,v 1.22 2006/11/28 16:59:56 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <inttypes.h>
+#include <string.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/oki_adpcm.h"
+
+/* Routines to convert 12 bit linear samples to the Oki ADPCM coding format,
+   widely used in CTI, because Dialogic use it. */
+
+static const int16_t step_size[49] =
+{
+      16,   17,   19,   21,   23,   25,   28,   31,
+      34,   37,   41,   45,   50,   55,   60,   66,
+      73,   80,   88,   97,  107,  118,  130,  143,
+     157,  173,  190,  209,  230,  253,  279,  307,
+     337,  371,  408,  449,  494,  544,  598,  658,
+     724,  796,  876,  963, 1060, 1166, 1282, 1408,
+    1552
+};
+
+static const int16_t step_adjustment[8] =
+{
+    -1, -1, -1, -1, 2, 4, 6, 8
+};
+
+/* Band limiting filter, to allow sample rate conversion to and
+   from 6k samples/second. */
+static const float cutoff_coeffs[] =
+{
+    -3.648392e-4f,
+     5.062391e-4f,
+     1.206247e-3f,
+     1.804452e-3f,
+     1.691750e-3f,
+     4.083405e-4f,
+    -1.931085e-3f,
+    -4.452107e-3f,
+    -5.794821e-3f,
+    -4.778489e-3f,
+    -1.161266e-3f,
+     3.928504e-3f,
+     8.259786e-3f,
+     9.500425e-3f,
+     6.512800e-3f,
+     2.227856e-4f,
+    -6.531275e-3f,
+    -1.026843e-2f,
+    -8.718062e-3f,
+    -2.280487e-3f,
+     5.817733e-3f,
+     1.096777e-2f,
+     9.634404e-3f,
+     1.569301e-3f,
+    -9.522632e-3f,
+    -1.748273e-2f,
+    -1.684408e-2f,
+    -6.100054e-3f,
+     1.071206e-2f,
+     2.525209e-2f,
+     2.871779e-2f,
+     1.664411e-2f,
+    -7.706268e-3f,
+    -3.331083e-2f,
+    -4.521249e-2f,
+    -3.085962e-2f,
+     1.373653e-2f,
+     8.089593e-2f,
+     1.529060e-1f,
+     2.080487e-1f,
+     2.286834e-1f,
+     2.080487e-1f,
+     1.529060e-1f,
+     8.089593e-2f,
+     1.373653e-2f,
+    -3.085962e-2f,
+    -4.521249e-2f,
+    -3.331083e-2f,
+    -7.706268e-3f,
+     1.664411e-2f,
+     2.871779e-2f,
+     2.525209e-2f,
+     1.071206e-2f,
+    -6.100054e-3f,
+    -1.684408e-2f,
+    -1.748273e-2f,
+    -9.522632e-3f,
+     1.569301e-3f,
+     9.634404e-3f,
+     1.096777e-2f,
+     5.817733e-3f,
+    -2.280487e-3f,
+    -8.718062e-3f,
+    -1.026843e-2f,
+    -6.531275e-3f,
+     2.227856e-4f,
+     6.512800e-3f,
+     9.500425e-3f,
+     8.259786e-3f,
+     3.928504e-3f,
+    -1.161266e-3f,
+    -4.778489e-3f,
+    -5.794821e-3f,
+    -4.452107e-3f,
+    -1.931085e-3f,
+     4.083405e-4f,
+     1.691750e-3f,
+     1.804452e-3f,
+     1.206247e-3f,
+     5.062391e-4f,
+    -3.648392e-4f
+};
+
+static int16_t decode(oki_adpcm_state_t *s, uint8_t adpcm)
+{
+    int16_t e;
+    int16_t ss;
+    int16_t linear;
+
+    /* Doing the next part as follows:
+     *
+     * x = adpcm & 0x07;
+     * e = (step_size[s->step_index]*(x + x + 1)) >> 3;
+     * 
+     * Seems an obvious improvement on a modern machine, but remember
+     * the truncation errors do not come out the same. It would
+     * not, therefore, be an exact match for what this code is doing.
+     *
+     * Just what a Dialogic card does, I do not know!
+     */
+
+    ss = step_size[s->step_index];
+    e = ss >> 3;
+    if (adpcm & 0x01)
+        e += (ss >> 2);
+    /*endif*/
+    if (adpcm & 0x02)
+        e += (ss >> 1);
+    /*endif*/
+    if (adpcm & 0x04)
+        e += ss;
+    /*endif*/
+    if (adpcm & 0x08)
+        e = -e;
+    /*endif*/
+    linear = s->last + e;
+
+    /* Saturate the values to +/- 2^11 (supposed to be 12 bits) */
+    if (linear > 2047)
+        linear = 2047;
+    else if (linear < -2048)
+        linear = -2048;
+    /*endif*/
+
+    s->last = linear;
+    s->step_index += step_adjustment[adpcm & 0x07];
+    if (s->step_index < 0)
+        s->step_index = 0;
+    else if (s->step_index > 48)
+        s->step_index = 48;
+    /*endif*/
+    /* Note: the result here is a 12 bit value */
+    return linear;
+}
+/*- End of function --------------------------------------------------------*/
+
+static uint8_t encode(oki_adpcm_state_t *s, int16_t linear)
+{
+    int16_t e;
+    int16_t ss;
+    uint8_t adpcm;
+
+    ss = step_size[s->step_index];
+    e = (linear >> 4) - s->last;
+    adpcm = (uint8_t) 0x00;
+    if (e < 0)
+    {
+        adpcm = (uint8_t) 0x08;
+        e = -e;
+    }
+    /*endif*/
+    if (e >= ss)
+    {
+        adpcm |= (uint8_t) 0x04;
+        e -= ss;
+    }
+    /*endif*/
+    if (e >= (ss >> 1))
+    {
+        adpcm |= (uint8_t) 0x02;
+        e -= ss;
+    }
+    /*endif*/
+    if (e >= (ss >> 2))
+        adpcm |= (uint8_t) 0x01;
+    /*endif*/
+
+    /* Use the decoder to set the estimate of the last sample. */
+    /* It also will adjust the step_index for us. */
+    s->last = decode(s, adpcm);
+    return adpcm;
+}
+/*- End of function --------------------------------------------------------*/
+
+oki_adpcm_state_t *oki_adpcm_init(oki_adpcm_state_t *s, int bit_rate)
+{
+    if (bit_rate != 32000  &&  bit_rate != 24000)
+        return NULL;
+    if (s == NULL)
+    {
+        if ((s = (oki_adpcm_state_t *) malloc(sizeof(*s))) == NULL)
+            return  NULL;
+    }
+    memset(s, 0, sizeof(*s));
+    s->bit_rate = bit_rate;
+    
+    return  s;
+}
+/*- End of function --------------------------------------------------------*/
+
+int oki_adpcm_release(oki_adpcm_state_t *s)
+{
+    free(s);
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+int oki_adpcm_decode(oki_adpcm_state_t *s,
+                     int16_t amp[],
+                     const uint8_t oki_data[],
+                     int oki_bytes)
+{
+    int i;
+    int x;
+    int l;
+    int n;
+    int samples;
+    float z;
+
+    samples = 0;
+    if (s->bit_rate == 32000)
+    {
+        for (i = 0;  i < oki_bytes;  i++)
+        {
+            amp[samples++] = decode(s, (oki_data[i] >> 4) & 0xF) << 4;
+            amp[samples++] = decode(s, oki_data[i] & 0xF) << 4;
+        }
+        /*endwhile*/
+    }
+    else
+    {
+        n = 0;
+        for (i = 0;  i < oki_bytes;  )
+        {
+            /* 6k to 8k sample/second conversion */
+            if (s->phase)
+            {
+                s->history[s->ptr++] =
+                    decode(s, (n++ & 1)  ?  (oki_data[i++] & 0xF)  :  ((oki_data[i] >> 4) & 0xF)) << 4;
+                s->ptr &= (32 - 1);
+            }
+            /*endif*/
+            z = 0.0f;
+            for (l = 80 - 3 + s->phase, x = s->ptr - 1;  l >= 0;  l -= 4, x--)
+                z += cutoff_coeffs[l]*s->history[x & (32 - 1)];
+            amp[samples++] = (int16_t) (z*4.0f);
+            if (++s->phase > 3)
+                s->phase = 0;
+            /*endif*/
+        }
+        /*endfor*/
+    }
+    /*endif*/
+    return  samples;
+}
+/*- End of function --------------------------------------------------------*/
+
+int oki_adpcm_encode(oki_adpcm_state_t *s,
+                     uint8_t oki_data[],
+                     const int16_t amp[],
+                     int len)
+{
+    int x;
+    int l;
+    int n;
+    int bytes;
+    float z;
+
+    bytes = 0;
+    if (s->bit_rate == 32000)
+    {
+        for (n = 0;  n < len;  n++)
+        {
+            s->oki_byte = (s->oki_byte << 4) | encode(s, amp[n]);
+            if ((s->mark++ & 1))
+                oki_data[bytes++] = s->oki_byte;
+            /*endif*/
+        }
+        /*endfor*/
+    }
+    else
+    {
+        n = 0;
+        for (;;)
+        {
+            /* 8k to 6k sample/second conversion */
+            if (s->phase > 2)
+            {
+                s->history[s->ptr++] = amp[n];
+                s->ptr &= (32 - 1);
+                s->phase = 0;
+                if (++n >= len)
+                    break;
+                /*endif*/
+            }
+            /*endif*/
+            s->history[s->ptr++] = amp[n];
+            s->ptr &= (32 - 1);
+            z = 0.0f;
+            for (l = 80 - s->phase, x = s->ptr - 1;  l >= 0;  l -= 3, x--)
+                z += cutoff_coeffs[l]*s->history[x & (32 - 1)];
+            /*endfor*/
+            s->oki_byte = (s->oki_byte << 4) | encode(s, (int16_t) (z*3.0f));
+            if ((s->mark++ & 1))
+                oki_data[bytes++] = s->oki_byte;
+            /*endif*/
+            s->phase++;
+            if (++n >= len)
+                break;
+            /*endif*/
+        }
+        /*endfor*/
+    }
+    /*endif*/
+    return  bytes;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/vector_int.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/vector_int.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,347 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * vector_int.c - Integer vector arithmetic
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser Lesser GNU General Public License version 2.1.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: vector_int.c,v 1.6 2007/08/21 14:25:54 steveu Exp $
+ */
+
+/*! \file */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <inttypes.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#if defined(HAVE_TGMATH_H)
+#include <tgmath.h>
+#endif
+#if defined(HAVE_MATH_H)
+#include <math.h>
+#endif
+#include <assert.h>
+
+#include "voipcodecs/telephony.h"
+#include "voipcodecs/vector_int.h"
+
+int32_t vec_dot_prodi16(const int16_t x[], const int16_t y[], int n)
+{
+    int32_t z;
+
+#if defined(__GNUC__)  &&  defined(__i386__)
+    __asm__ __volatile__(
+        " emms;\n"
+        " pxor %%mm0,%%mm0;\n"
+        " leal -32(%%esi,%%eax,2),%%edx;\n"     /* edx = top - 32 */
+
+        " cmpl %%edx,%%esi;\n"
+        " ja 1f;\n"
+
+        /* Work in blocks of 16 int16_t's until we are near the end */
+        " .p2align 2;\n"
+        "2:\n"
+        " movq (%%edi),%%mm1;\n"
+        " movq (%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq 8(%%edi),%%mm1;\n"
+        " movq 8(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq 16(%%edi),%%mm1;\n"
+        " movq 16(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        " movq 24(%%edi),%%mm1;\n"
+        " movq 24(%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+
+        " addl $32,%%esi;\n"
+        " addl $32,%%edi;\n"
+        " cmpl %%edx,%%esi;\n"
+        " jbe 2b;\n"
+
+        " .p2align 2;\n"
+        "1:\n"
+        " addl $24,%%edx;\n"                  /* now edx = top - 8 */
+        " cmpl %%edx,%%esi;\n"
+        " ja 3f;\n"
+
+        /* Work in blocks of 4 int16_t's until we are near the end */
+        " .p2align 2;\n"
+        "4:\n"
+        " movq (%%edi),%%mm1;\n"
+        " movq (%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+
+        " addl $8,%%esi;\n"
+        " addl $8,%%edi;\n"
+        " cmpl %%edx,%%esi;"
+        " jbe 4b;\n"
+
+        " .p2align 2;\n"
+        "3:\n"
+        " addl $4,%%edx;\n"                  /* now edx = top - 4 */
+        " cmpl %%edx,%%esi;\n"
+        " ja 5f;\n"
+
+        /* Work in a block of 2 int16_t's */
+        " movd (%%edi),%%mm1;\n"
+        " movd (%%esi),%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+
+        " addl $4,%%esi;\n"
+        " addl $4,%%edi;\n"
+
+        " .p2align 2;\n"
+        "5:\n"
+        " addl $2,%%edx;\n"                  /* now edx = top - 2 */
+        " cmpl %%edx,%%esi;\n"
+        " ja 6f;\n"
+
+        /* Deal with the very last int16_t, when n is odd */
+        " movswl (%%edi),%%eax;\n"
+        " andl $65535,%%eax;\n"
+        " movd %%eax,%%mm1;\n"
+        " movswl (%%esi),%%eax;\n"
+        " andl $65535,%%eax;\n"
+        " movd %%eax,%%mm2;\n"
+        " pmaddwd %%mm2,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+
+        " .p2align 2;\n"
+        "6:\n"
+        /* Merge the pieces of the answer */
+        " movq %%mm0,%%mm1;\n"
+        " punpckhdq %%mm0,%%mm1;\n"
+        " paddd %%mm1,%%mm0;\n"
+        /* et voila, eax has the final result */
+        " movd %%mm0,%%eax;\n"
+
+        " emms;\n"
+        : "=a" (z)
+        : "S" (x), "D" (y), "a" (n)
+        : "cc"
+    );
+#else
+    int i;
+
+    z = 0;
+    for (i = 0;  i < n;  i++)
+        z += (int32_t) x[i]*(int32_t) y[i];
+#endif
+    return  z;
+}
+/*- End of function --------------------------------------------------------*/
+
+int32_t vec_min_maxi16(const int16_t x[], int n, int16_t out[])
+{
+#if defined(__GNUC__)  &&  defined(__i386__)
+    static const int32_t lower_bound = 0x80008000;
+    static const int32_t upper_bound = 0x7FFF7FFF;
+    int32_t max;
+
+    __asm__ __volatile__(
+        " emms;\n"
+        " pushl %%edx;\n"
+        " leal -8(%%esi,%%eax,2),%%edx;\n"
+
+        " cmpl %%edx,%%esi;\n"
+        " jbe 2f;\n"
+        " movd %[lower],%%mm0;\n"
+        " movd %[upper],%%mm1;\n"
+        " jmp 1f;\n"
+
+        " .p2align 2;\n"
+        "2:\n"
+        " movq (%%esi),%%mm0;\n"   /* mm0 will be max's */
+        " movq %%mm0,%%mm1;\n"     /* mm1 will be min's */
+        " addl $8,%%esi;\n"
+        " cmpl %%edx,%%esi;\n"
+        " ja 4f;\n"
+
+        " .p2align 2;\n"
+        "3:\n"
+        " movq (%%esi),%%mm2;\n"
+
+        " movq %%mm2,%%mm3;\n"
+        " pcmpgtw %%mm0,%%mm3;\n"  /* mm3 is bitmask for words where mm2 > mm0 */ 
+        " movq %%mm3,%%mm4;\n"
+        " pand %%mm2,%%mm3;\n"     /* mm3 is mm2 masked to new max's */
+        " pandn %%mm0,%%mm4;\n"    /* mm4 is mm0 masked to its max's */
+        " por %%mm3,%%mm4;\n"
+        " movq %%mm4,%%mm0;\n"     /* Now mm0 is updated max's */
+        
+        " movq %%mm1,%%mm3;\n"
+        " pcmpgtw %%mm2,%%mm3;\n"  /* mm3 is bitmask for words where mm2 < mm1 */ 
+        " pand %%mm3,%%mm2;\n"     /* mm2 is mm2 masked to new min's */
+        " pandn %%mm1,%%mm3;\n"    /* mm3 is mm1 masked to its min's */
+        " por %%mm3,%%mm2;\n"
+        " movq %%mm2,%%mm1;\n"     /* now mm1 is updated min's */
+
+        " addl $8,%%esi;\n"
+        " cmpl %%edx,%%esi;\n"
+        " jbe 3b;\n"
+
+        " .p2align 2;\n"
+        "4:\n"
+        /* Merge down the 4-word max/mins to lower 2 words */
+        " movq %%mm0,%%mm2;\n"
+        " psrlq $32,%%mm2;\n"
+        " movq %%mm2,%%mm3;\n"
+        " pcmpgtw %%mm0,%%mm3;\n"  /* mm3 is bitmask for words where mm2 > mm0 */ 
+        " pand %%mm3,%%mm2;\n"     /* mm2 is mm2 masked to new max's */
+        " pandn %%mm0,%%mm3;\n"    /* mm3 is mm0 masked to its max's */
+        " por %%mm3,%%mm2;\n"
+        " movq %%mm2,%%mm0;\n"     /* now mm0 is updated max's */
+
+        " movq %%mm1,%%mm2;\n"
+        " psrlq $32,%%mm2;\n"
+        " movq %%mm1,%%mm3;\n"
+        " pcmpgtw %%mm2,%%mm3;\n"  /* mm3 is bitmask for words where mm2 < mm1 */ 
+        " pand %%mm3,%%mm2;\n"     /* mm2 is mm2 masked to new min's */
+        " pandn %%mm1,%%mm3;\n"    /* mm3 is mm1 masked to its min's */
+        " por %%mm3,%%mm2;\n"
+        " movq %%mm2,%%mm1;\n"     /* now mm1 is updated min's */
+
+        " .p2align 2;\n"
+        "1:\n"
+        " addl $4,%%edx;\n"        /* now dx = top-4 */
+        " cmpl %%edx,%%esi;\n"
+        " ja 5f;\n"
+        /* Here, there are >= 2 words of input remaining */
+        " movd (%%esi),%%mm2;\n"
+
+        " movq %%mm2,%%mm3;\n"
+        " pcmpgtw %%mm0,%%mm3;\n"  /* mm3 is bitmask for words where mm2 > mm0 */ 
+        " movq %%mm3,%%mm4;\n"
+        " pand %%mm2,%%mm3;\n"     /* mm3 is mm2 masked to new max's */
+        " pandn %%mm0,%%mm4;\n"    /* mm4 is mm0 masked to its max's */
+        " por %%mm3,%%mm4;\n"
+        " movq %%mm4,%%mm0;\n"     /* now mm0 is updated max's */
+
+        " movq %%mm1,%%mm3;\n"
+        " pcmpgtw %%mm2,%%mm3;\n"  /* mm3 is bitmask for words where mm2 < mm1 */ 
+        " pand %%mm3,%%mm2;\n"     /* mm2 is mm2 masked to new min's */
+        " pandn %%mm1,%%mm3;\n"    /* mm3 is mm1 masked to its min's */
+        " por %%mm3,%%mm2;\n"
+        " movq %%mm2,%%mm1;\n"     /* now mm1 is updated min's */
+
+        " addl $4,%%esi;\n"
+
+        " .p2align 2;\n"
+        "5:\n"
+        /* Merge down the 2-word max/mins to 1 word */
+        " movq %%mm0,%%mm2;\n"
+        " psrlq $16,%%mm2;\n"
+        " movq %%mm2,%%mm3;\n"
+        " pcmpgtw %%mm0,%%mm3;\n"  /* mm3 is bitmask for words where mm2 > mm0 */ 
+        " pand %%mm3,%%mm2;\n"     /* mm2 is mm2 masked to new max's */
+        " pandn %%mm0,%%mm3;\n"    /* mm3 is mm0 masked to its max's */
+        " por %%mm3,%%mm2;\n"
+        " movd %%mm2,%%ecx;\n"     /* cx is max so far */
+
+        " movq %%mm1,%%mm2;\n"
+        " psrlq $16,%%mm2;\n"
+        " movq %%mm1,%%mm3;\n"
+        " pcmpgtw %%mm2,%%mm3;\n"  /* mm3 is bitmask for words where mm2 < mm1 */ 
+        " pand %%mm3,%%mm2;\n"     /* mm2 is mm2 masked to new min's */
+        " pandn %%mm1,%%mm3;\n"    /* mm3 is mm1 masked to its min's */
+        " por %%mm3,%%mm2;\n"
+        " movd %%mm2,%%eax;\n"     /* ax is min so far */
+        
+        " addl $2,%%edx;\n"        /* now dx = top-2 */
+        " cmpl %%edx,%%esi;\n"
+        " ja 6f;\n"
+
+        /* Here, there is one word of input left */
+        " cmpw (%%esi),%%cx;\n"
+        " jge 9f;\n"
+        " movw (%%esi),%%cx;\n"
+        " .p2align 2;\n"
+        "9:\n"
+        " cmpw (%%esi),%%ax;\n"
+        " jle 6f;\n"
+        " movw (%%esi),%%ax;\n"
+
+        " .p2align 2;\n"
+        "6:\n"
+        /* (finally!) cx is the max, ax the min */
+        " movswl %%cx,%%ecx;\n"
+        " movswl %%ax,%%eax;\n"
+
+        " popl %%edx;\n"            /* ptr to output max,min vals */
+        " andl %%edx,%%edx;\n"
+        " jz 7f;\n"
+        " movw %%cx,(%%edx);\n"    /* max */
+        " movw %%ax,2(%%edx);\n"   /* min */
+        " .p2align 2;\n"
+        "7:\n"
+        /* Now calculate max absolute value */
+        " negl %%eax;\n"
+        " cmpl %%ecx,%%eax;\n"
+        " jge 8f;\n"
+        " movl %%ecx,%%eax;\n"
+        " .p2align 2;\n"
+        "8:\n"
+        " emms;\n"
+        : "=a" (max)
+        : "S" (x), "a" (n), "d" (out), [lower] "m" (lower_bound), [upper] "m" (upper_bound)
+        : "ecx"
+    );
+    return  max;
+#else
+    int i;
+    int16_t min;
+    int16_t max;
+    int16_t temp;
+    int32_t z;
+    
+    max = INT16_MIN;
+    min = INT16_MAX;
+    for (i = 0;  i < n;  i++)
+    {
+        temp = x[i];
+        if (temp > max)
+            max = temp;
+        /*endif*/
+        if (temp < min)
+            min = temp;
+        /*endif*/
+    }
+    /*endfor*/
+    out[0] = max;
+    out[1] = min;
+    z = abs(min);
+    if (z > max)
+        return z;
+    return max;
+#endif
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs.h.in
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs.h.in	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,54 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * voipcodecs.h - The head guy amongst the headers
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2003 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id$
+ */
+
+/*! \file */
+
+#if !defined(_VOIPCODECS_H_)
+#define _VOIPCODECS_H_
+
+@VOIPCODECS_USE_FIXED_POINT@
+
+#include <stdlib.h>
+@INSERT_INTTYPES_HEADER@
+#include <string.h>
+#include <limits.h>
+#include <time.h>
+@INSERT_MATH_HEADER@
+
+#include <voipcodecs/telephony.h>
+#include <voipcodecs/bit_operations.h>
+#include <voipcodecs/bitstream.h>
+#include <voipcodecs/g711.h>
+#include <voipcodecs/g722.h>
+#include <voipcodecs/g726.h>
+#include <voipcodecs/gsm0610.h>
+#include <voipcodecs/ima_adpcm.h>
+#include <voipcodecs/lpc10.h>
+#include <voipcodecs/oki_adpcm.h>
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/bit_operations.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/bit_operations.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,262 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * bit_operations.h - Various bit level operations, such as bit reversal
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: bit_operations.h,v 1.19 2007/12/13 11:31:32 steveu Exp $
+ */
+
+/*! \file */
+
+#if !defined(_SPANDSP_BIT_OPERATIONS_H_)
+#define _SPANDSP_BIT_OPERATIONS_H_
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+/*! \brief Find the bit position of the highest set bit in a word
+    \param bits The word to be searched
+    \return The bit number of the highest set bit, or -1 if the word is zero. */
+static __inline__ int top_bit(unsigned int bits)
+{
+    int res;
+    
+#if defined(__i386__)  ||  defined(__x86_64__)
+    __asm__ (" xorl %[res],%[res];\n"
+             " decl %[res];\n"
+             " bsrl %[bits],%[res]\n"
+             : [res] "=&r" (res)
+             : [bits] "rm" (bits));
+    return res;
+#elif defined(__ppc__)  ||   defined(__powerpc__)
+    __asm__ ("cntlzw %[res],%[bits];\n"
+             : [res] "=&r" (res)
+             : [bits] "r" (bits));
+    return 31 - res;
+#else
+    if (bits == 0)
+        return -1;
+    res = 0;
+    if (bits & 0xFFFF0000)
+    {
+        bits &= 0xFFFF0000;
+        res += 16;
+    }
+    if (bits & 0xFF00FF00)
+    {
+        bits &= 0xFF00FF00;
+        res += 8;
+    }
+    if (bits & 0xF0F0F0F0)
+    {
+        bits &= 0xF0F0F0F0;
+        res += 4;
+    }
+    if (bits & 0xCCCCCCCC)
+    {
+        bits &= 0xCCCCCCCC;
+        res += 2;
+    }
+    if (bits & 0xAAAAAAAA)
+    {
+        bits &= 0xAAAAAAAA;
+        res += 1;
+    }
+    return res;
+#endif
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Find the bit position of the lowest set bit in a word
+    \param bits The word to be searched
+    \return The bit number of the lowest set bit, or -1 if the word is zero. */
+static __inline__ int bottom_bit(unsigned int bits)
+{
+    int res;
+    
+#if defined(__i386__)  ||  defined(__x86_64__)
+    __asm__ (" xorl %[res],%[res];\n"
+             " decl %[res];\n"
+             " bsfl %[bits],%[res]\n"
+             : [res] "=&r" (res)
+             : [bits] "rm" (bits));
+    return res;
+#else
+    if (bits == 0)
+        return -1;
+    res = 31;
+    if (bits & 0x0000FFFF)
+    {
+        bits &= 0x0000FFFF;
+        res -= 16;
+    }
+    if (bits & 0x00FF00FF)
+    {
+        bits &= 0x00FF00FF;
+        res -= 8;
+    }
+    if (bits & 0x0F0F0F0F)
+    {
+        bits &= 0x0F0F0F0F;
+        res -= 4;
+    }
+    if (bits & 0x33333333)
+    {
+        bits &= 0x33333333;
+        res -= 2;
+    }
+    if (bits & 0x55555555)
+    {
+        bits &= 0x55555555;
+        res -= 1;
+    }
+    return res;
+#endif
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Bit reverse a byte.
+    \param data The byte to be reversed.
+    \return The bit reversed version of data. */
+static __inline__ uint8_t bit_reverse8(uint8_t x)
+{
+#if defined(__i386__)  ||  defined(__x86_64__)  ||  defined(__ppc__)  ||  defined(__powerpc__)
+    /* If multiply is fast */
+    return ((x*0x0802U & 0x22110U) | (x*0x8020U & 0x88440U))*0x10101U >> 16;
+#else
+    /* If multiply is slow, but we have a barrel shifter */
+    x = (x >> 4) | (x << 4);
+    x = ((x & 0xCC) >> 2) | ((x & 0x33) << 2);
+    return ((x & 0xAA) >> 1) | ((x & 0x55) << 1);
+#endif
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Bit reverse a 16 bit word.
+    \param data The word to be reversed.
+    \return The bit reversed version of data. */
+uint16_t bit_reverse16(uint16_t data);
+
+/*! \brief Bit reverse a 32 bit word.
+    \param data The word to be reversed.
+    \return The bit reversed version of data. */
+uint32_t bit_reverse32(uint32_t data);
+
+/*! \brief Bit reverse each of the four bytes in a 32 bit word.
+    \param data The word to be reversed.
+    \return The bit reversed version of data. */
+uint32_t bit_reverse_4bytes(uint32_t data);
+
+#if defined(__x86_64__)
+/*! \brief Bit reverse each of the eight bytes in a 64 bit word.
+    \param data The word to be reversed.
+    \return The bit reversed version of data. */
+uint64_t bit_reverse_8bytes(uint64_t data);
+#endif
+
+/*! \brief Bit reverse each bytes in a buffer.
+    \param to The buffer to place the reversed data in.
+    \param from The buffer containing the data to be reversed.
+    \param len The length of the data in the buffer. */
+void bit_reverse(uint8_t to[], const uint8_t from[], int len);
+
+/*! \brief Find the number of set bits in a 32 bit word.
+    \param x The word to be searched.
+    \return The number of set bits. */
+int one_bits32(uint32_t x);
+
+/*! \brief Create a mask as wide as the number in a 32 bit word.
+    \param x The word to be searched.
+    \return The mask. */
+uint32_t make_mask32(uint32_t x);
+
+/*! \brief Create a mask as wide as the number in a 16 bit word.
+    \param x The word to be searched.
+    \return The mask. */
+uint16_t make_mask16(uint16_t x);
+
+/*! \brief Find the least significant one in a word, and return a word
+           with just that bit set.
+    \param x The word to be searched.
+    \return The word with the single set bit. */
+static __inline__ uint32_t least_significant_one32(uint32_t x)
+{
+    return (x & (-(int32_t) x));
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Find the most significant one in a word, and return a word
+           with just that bit set.
+    \param x The word to be searched.
+    \return The word with the single set bit. */
+static __inline__ uint32_t most_significant_one32(uint32_t x)
+{
+#if defined(__i386__)  ||  defined(__x86_64__)  ||  defined(__ppc__)  ||  defined(__powerpc__)
+    return 1 << top_bit(x);
+#else
+    x = make_mask32(x);
+    return (x ^ (x >> 1));
+#endif
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Find the parity of a byte.
+    \param x The byte to be checked.
+    \return 1 for odd, or 0 for even. */
+static __inline__ int parity8(uint8_t x)
+{
+    x = (x ^ (x >> 4)) & 0x0F;
+    return (0x6996 >> x) & 1;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Find the parity of a 16 bit word.
+    \param x The word to be checked.
+    \return 1 for odd, or 0 for even. */
+static __inline__ int parity16(uint16_t x)
+{
+    x ^= (x >> 8);
+    x = (x ^ (x >> 4)) & 0x0F;
+    return (0x6996 >> x) & 1;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Find the parity of a 32 bit word.
+    \param x The word to be checked.
+    \return 1 for odd, or 0 for even. */
+static __inline__ int parity32(uint32_t x)
+{
+    x ^= (x >> 16);
+    x ^= (x >> 8);
+    x = (x ^ (x >> 4)) & 0x0F;
+    return (0x6996 >> x) & 1;
+}
+/*- End of function --------------------------------------------------------*/
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/bitstream.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/bitstream.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,89 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * bitstream.h - Bitstream composition and decomposition routines.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: bitstream.h,v 1.8 2007/12/13 11:31:32 steveu Exp $
+ */
+
+/*! \file */
+
+#if !defined(_SPANDSP_BITSTREAM_H_)
+#define _SPANDSP_BITSTREAM_H_
+
+/*! \page bitstream_page Bitstream composition and decomposition
+\section bitstream_page_sec_1 What does it do?
+
+\section bitstream_page_sec_2 How does it work?
+*/
+
+/*! Bitstream handler state */
+typedef struct
+{
+    /*! The bit stream. */
+    unsigned int bitstream;
+    /*! The residual bits in bitstream. */
+    unsigned int residue;
+} bitstream_state_t;
+
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+/*! \brief Put a chunk of bits into the output buffer.
+    \param s A pointer to the bitstream context.
+    \param c A pointer to the bitstream output buffer.
+    \param value The value to be pushed into the output buffer.
+    \param bits The number of bits of value to be pushed. 1 to 25 bit is valid. */
+void bitstream_put(bitstream_state_t *s, uint8_t **c, unsigned int value, int bits);
+
+void bitstream_put2(bitstream_state_t *s, uint8_t **c, unsigned int value, int bits);
+
+/*! \brief Get a chunk of bits from the input buffer.
+    \param s A pointer to the bitstream context.
+    \param c A pointer to the bitstream input buffer.
+    \param bits The number of bits of value to be grabbed. 1 to 25 bit is valid.
+    \return The value retrieved from the input buffer. */
+unsigned int bitstream_get(bitstream_state_t *s, const uint8_t **c, int bits);
+
+unsigned int bitstream_get2(bitstream_state_t *s, const uint8_t **c, int bits);
+
+/*! \brief Flush any residual bit to the output buffer.
+    \param s A pointer to the bitstream context.
+    \param c A pointer to the bitstream output buffer. */
+void bitstream_flush(bitstream_state_t *s, uint8_t **c);
+
+void bitstream_flush2(bitstream_state_t *s, uint8_t **c);
+
+/*! \brief Initialise a bitstream context.
+    \param s A pointer to the bitstream context.
+    \return A pointer to the bitstream context. */
+bitstream_state_t *bitstream_init(bitstream_state_t *s);
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/dc_restore.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/dc_restore.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,127 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * dc_restore.h - General telephony routines to restore the zero D.C.
+ *                level to audio which has a D.C. bias.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2001 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: dc_restore.h,v 1.18 2007/04/08 08:16:17 steveu Exp $
+ */
+
+/*! \file */
+
+#if !defined(_SPANDSP_DC_RESTORE_H_)
+#define _SPANDSP_DC_RESTORE_H_
+
+/*! \page dc_restore_page Removing DC bias from a signal
+
+\section dc_restore_page_sec_1 What does it do?
+
+Telecoms signals often contain considerable DC, but DC upsets a lot of signal
+processing functions. Placing a zero DC restorer at the front of the processing
+chain can often simplify the downstream processing. 
+
+\section dc_restore_page_sec_2 How does it work?
+
+The DC restorer uses a leaky integrator to provide a long-ish term estimate of
+the DC bias in the signal. A 32 bit estimate is used for the 16 bit audio, so
+the noise introduced by the estimation can be keep in the lower bits, and the 16
+bit DC value, which is subtracted from the signal, is fairly clean. The
+following code fragment shows the algorithm used. dc_bias is a 32 bit integer,
+while the sample and the resulting clean_sample are 16 bit integers. 
+
+    dc_bias += ((((int32_t) sample << 15) - dc_bias) >> 14);
+    clean_sample = sample - (dc_bias >> 15); 
+*/
+
+/*!
+    Zero DC restoration descriptor. This defines the working state for a single
+    instance of DC content filter.
+*/
+typedef struct
+{
+    int32_t state;
+} dc_restore_state_t;
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+static __inline__ void dc_restore_init(dc_restore_state_t *dc)
+{
+    dc->state = 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int16_t dc_restore(dc_restore_state_t *dc, int16_t sample)
+{
+    dc->state += ((((int32_t) sample << 15) - dc->state) >> 14);
+    return (int16_t) (sample - (dc->state >> 15));
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int16_t dc_restore_estimate(dc_restore_state_t *dc)
+{
+    return (int16_t) (dc->state >> 15);
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int16_t saturate(int32_t amp)
+{
+    int16_t amp16;
+
+    /* Hopefully this is optimised for the common case - not clipping */
+    amp16 = (int16_t) amp;
+    if (amp == amp16)
+        return amp16;
+    if (amp > INT16_MAX)
+        return  INT16_MAX;
+    return  INT16_MIN;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int16_t fsaturatef(float famp)
+{
+    if (famp > 32767.0)
+        return  INT16_MAX;
+    if (famp < -32768.0)
+        return  INT16_MIN;
+    return (int16_t) rintf(famp);
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int16_t fsaturate(double damp)
+{
+    if (damp > 32767.0)
+        return  INT16_MAX;
+    if (damp < -32768.0)
+        return  INT16_MIN;
+    return (int16_t) rint(damp);
+}
+/*- End of function --------------------------------------------------------*/
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/g711.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/g711.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,255 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g711.h - In line A-law and u-law conversion routines
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2001 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: g711.h,v 1.8 2007/12/13 11:31:32 steveu Exp $
+ */
+
+/*! \file */
+
+/*! \page g711_page A-law and mu-law handling
+Lookup tables for A-law and u-law look attractive, until you consider the impact
+on the CPU cache. If it causes a substantial area of your processor cache to get
+hit too often, cache sloshing will severely slow things down. The main reason
+these routines are slow in C, is the lack of direct access to the CPU's "find
+the first 1" instruction. A little in-line assembler fixes that, and the
+conversion routines can be faster than lookup tables, in most real world usage.
+A "find the first 1" instruction is available on most modern CPUs, and is a
+much underused feature. 
+
+If an assembly language method of bit searching is not available, these routines
+revert to a method that can be a little slow, so the cache thrashing might not
+seem so bad :(
+
+Feel free to submit patches to add fast "find the first 1" support for your own
+favourite processor.
+
+Look up tables are used for transcoding between A-law and u-law, since it is
+difficult to achieve the precise transcoding procedure laid down in the G.711
+specification by other means.
+*/
+
+#if !defined(_SPANDSP_G711_H_)
+#define _SPANDSP_G711_H_
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+/* N.B. It is tempting to use look-up tables for A-law and u-law conversion.
+ *      However, you should consider the cache footprint.
+ *
+ *      A 64K byte table for linear to x-law and a 512 byte table for x-law to
+ *      linear sound like peanuts these days, and shouldn't an array lookup be
+ *      real fast? No! When the cache sloshes as badly as this one will, a tight
+ *      calculation may be better. The messiest part is normally finding the
+ *      segment, but a little inline assembly can fix that on an i386, x86_64 and
+ *      many other modern processors.
+ */
+ 
+/*
+ * Mu-law is basically as follows:
+ *
+ *      Biased Linear Input Code        Compressed Code
+ *      ------------------------        ---------------
+ *      00000001wxyza                   000wxyz
+ *      0000001wxyzab                   001wxyz
+ *      000001wxyzabc                   010wxyz
+ *      00001wxyzabcd                   011wxyz
+ *      0001wxyzabcde                   100wxyz
+ *      001wxyzabcdef                   101wxyz
+ *      01wxyzabcdefg                   110wxyz
+ *      1wxyzabcdefgh                   111wxyz
+ *
+ * Each biased linear code has a leading 1 which identifies the segment
+ * number. The value of the segment number is equal to 7 minus the number
+ * of leading 0's. The quantization interval is directly available as the
+ * four bits wxyz.  * The trailing bits (a - h) are ignored.
+ *
+ * Ordinarily the complement of the resulting code word is used for
+ * transmission, and so the code word is complemented before it is returned.
+ *
+ * For further information see John C. Bellamy's Digital Telephony, 1982,
+ * John Wiley & Sons, pps 98-111 and 472-476.
+ */
+
+//#define ULAW_ZEROTRAP                 /* turn on the trap as per the MIL-STD */
+#define ULAW_BIAS        0x84           /* Bias for linear code. */
+
+/*! \brief Encode a linear sample to u-law
+    \param linear The sample to encode.
+    \return The u-law value.
+*/
+static __inline__ uint8_t linear_to_ulaw(int linear)
+{
+    uint8_t u_val;
+    int mask;
+    int seg;
+
+    /* Get the sign and the magnitude of the value. */
+    if (linear >= 0)
+    {
+        linear = ULAW_BIAS + linear;
+        mask = 0xFF;
+    }
+    else
+    {
+        linear = ULAW_BIAS - linear;
+        mask = 0x7F;
+    }
+
+    seg = top_bit(linear | 0xFF) - 7;
+
+    /*
+     * Combine the sign, segment, quantization bits,
+     * and complement the code word.
+     */
+    if (seg >= 8)
+        u_val = (uint8_t) (0x7F ^ mask);
+    else
+        u_val = (uint8_t) (((seg << 4) | ((linear >> (seg + 3)) & 0xF)) ^ mask);
+#ifdef ULAW_ZEROTRAP
+    /* Optional ITU trap */
+    if (u_val == 0)
+        u_val = 0x02;
+#endif
+    return  u_val;
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Decode an u-law sample to a linear value.
+    \param ulaw The u-law sample to decode.
+    \return The linear value.
+*/
+static __inline__ int16_t ulaw_to_linear(uint8_t ulaw)
+{
+    int t;
+    
+    /* Complement to obtain normal u-law value. */
+    ulaw = ~ulaw;
+    /*
+     * Extract and bias the quantization bits. Then
+     * shift up by the segment number and subtract out the bias.
+     */
+    t = (((ulaw & 0x0F) << 3) + ULAW_BIAS) << (((int) ulaw & 0x70) >> 4);
+    return  (int16_t) ((ulaw & 0x80)  ?  (ULAW_BIAS - t)  :  (t - ULAW_BIAS));
+}
+/*- End of function --------------------------------------------------------*/
+
+/*
+ * A-law is basically as follows:
+ *
+ *      Linear Input Code        Compressed Code
+ *      -----------------        ---------------
+ *      0000000wxyza             000wxyz
+ *      0000001wxyza             001wxyz
+ *      000001wxyzab             010wxyz
+ *      00001wxyzabc             011wxyz
+ *      0001wxyzabcd             100wxyz
+ *      001wxyzabcde             101wxyz
+ *      01wxyzabcdef             110wxyz
+ *      1wxyzabcdefg             111wxyz
+ *
+ * For further information see John C. Bellamy's Digital Telephony, 1982,
+ * John Wiley & Sons, pps 98-111 and 472-476.
+ */
+
+#define ALAW_AMI_MASK       0x55
+
+/*! \brief Encode a linear sample to A-law
+    \param linear The sample to encode.
+    \return The A-law value.
+*/
+static __inline__ uint8_t linear_to_alaw(int linear)
+{
+    int mask;
+    int seg;
+    
+    if (linear >= 0)
+    {
+        /* Sign (bit 7) bit = 1 */
+        mask = ALAW_AMI_MASK | 0x80;
+    }
+    else
+    {
+        /* Sign (bit 7) bit = 0 */
+        mask = ALAW_AMI_MASK;
+        linear = -linear - 1;
+    }
+
+    /* Convert the scaled magnitude to segment number. */
+    seg = top_bit(linear | 0xFF) - 7;
+    if (seg >= 8)
+    {
+        if (linear >= 0)
+        {
+            /* Out of range. Return maximum value. */
+            return (uint8_t) (0x7F ^ mask);
+        }
+        /* We must be just a tiny step below zero */
+        return (uint8_t) (0x00 ^ mask);
+    }
+    /* Combine the sign, segment, and quantization bits. */
+    return (uint8_t) (((seg << 4) | ((linear >> ((seg)  ?  (seg + 3)  :  4)) & 0x0F)) ^ mask);
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Decode an A-law sample to a linear value.
+    \param alaw The A-law sample to decode.
+    \return The linear value.
+*/
+static __inline__ int16_t alaw_to_linear(uint8_t alaw)
+{
+    int i;
+    int seg;
+
+    alaw ^= ALAW_AMI_MASK;
+    i = ((alaw & 0x0F) << 4);
+    seg = (((int) alaw & 0x70) >> 4);
+    if (seg)
+        i = (i + 0x108) << (seg - 1);
+    else
+        i += 8;
+    return (int16_t) ((alaw & 0x80)  ?  i  :  -i);
+}
+/*- End of function --------------------------------------------------------*/
+
+/*! \brief Transcode from A-law to u-law, using the procedure defined in G.711.
+    \param alaw The A-law sample to transcode.
+    \return The best matching u-law value.
+*/
+uint8_t alaw_to_ulaw(uint8_t alaw);
+
+/*! \brief Transcode from u-law to A-law, using the procedure defined in G.711.
+    \param ulaw The u-law sample to transcode.
+    \return The best matching A-law value.
+*/
+uint8_t ulaw_to_alaw(uint8_t ulaw);
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/g722.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/g722.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,180 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g722.h - The ITU G.722 codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * Based on a single channel G.722 codec which is:
+ *
+ *****    Copyright (c) CMU    1993      *****
+ * Computer Science, Speech Group
+ * Chengxiang Lu and Alex Hauptmann
+ *
+ * $Id: g722.h,v 1.16 2007/04/08 08:16:17 steveu Exp $
+ */
+
+
+/*! \file */
+
+#if !defined(_SPANDSP_G722_H_)
+#define _SPANDSP_G722_H_
+
+/*! \page g722_page G.722 encoding and decoding
+\section g722_page_sec_1 What does it do?
+The G.722 module is a bit exact implementation of the ITU G.722 specification for all three
+specified bit rates - 64000bps, 56000bps and 48000bps. It passes the ITU tests.
+
+To allow fast and flexible interworking with narrow band telephony, the encoder and decoder
+support an option for the linear audio to be an 8k samples/second stream. In this mode the
+codec is considerably faster, and still fully compatible with wideband terminals using G.722.
+
+\section g722_page_sec_2 How does it work?
+???.
+*/
+
+enum
+{
+    G722_SAMPLE_RATE_8000 = 0x0001,
+    G722_PACKED = 0x0002
+};
+
+typedef struct
+{
+    /*! TRUE if the operating in the special ITU test mode, with the band split filters
+             disabled. */
+    int itu_test_mode;
+    /*! TRUE if the G.722 data is packed */
+    int packed;
+    /*! TRUE if encode from 8k samples/second */
+    int eight_k;
+    /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
+    int bits_per_sample;
+
+    /*! Signal history for the QMF */
+    int x[24];
+
+    struct
+    {
+        int s;
+        int sp;
+        int sz;
+        int r[3];
+        int a[3];
+        int ap[3];
+        int p[3];
+        int d[7];
+        int b[7];
+        int bp[7];
+        int sg[7];
+        int nb;
+        int det;
+    } band[2];
+
+    unsigned int in_buffer;
+    int in_bits;
+    unsigned int out_buffer;
+    int out_bits;
+} g722_encode_state_t;
+
+typedef struct
+{
+    /*! TRUE if the operating in the special ITU test mode, with the band split filters
+             disabled. */
+    int itu_test_mode;
+    /*! TRUE if the G.722 data is packed */
+    int packed;
+    /*! TRUE if decode to 8k samples/second */
+    int eight_k;
+    /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
+    int bits_per_sample;
+
+    /*! Signal history for the QMF */
+    int x[24];
+
+    struct
+    {
+        int s;
+        int sp;
+        int sz;
+        int r[3];
+        int a[3];
+        int ap[3];
+        int p[3];
+        int d[7];
+        int b[7];
+        int bp[7];
+        int sg[7];
+        int nb;
+        int det;
+    } band[2];
+    
+    unsigned int in_buffer;
+    int in_bits;
+    unsigned int out_buffer;
+    int out_bits;
+} g722_decode_state_t;
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+/*! Initialise an G.722 encode context.
+    \param s The G.722 encode context.
+    \param rate The required bit rate for the G.722 data.
+           The valid rates are 64000, 56000 and 48000.
+    \param options
+    \return A pointer to the G.722 encode context, or NULL for error. */
+g722_encode_state_t *g722_encode_init(g722_encode_state_t *s, int rate, int options);
+
+int g722_encode_release(g722_encode_state_t *s);
+
+/*! Encode a buffer of linear PCM data to G.722
+    \param s The G.722 context.
+    \param g722_data The G.722 data produced.
+    \param amp The audio sample buffer.
+    \param len The number of samples in the buffer.
+    \return The number of bytes of G.722 data produced. */
+int g722_encode(g722_encode_state_t *s, uint8_t g722_data[], const int16_t amp[], int len);
+
+/*! Initialise an G.722 decode context.
+    \param s The G.722 decode context.
+    \param rate The bit rate of the G.722 data.
+           The valid rates are 64000, 56000 and 48000.
+    \param options
+    \return A pointer to the G.722 decode context, or NULL for error. */
+g722_decode_state_t *g722_decode_init(g722_decode_state_t *s, int rate, int options);
+
+int g722_decode_release(g722_decode_state_t *s);
+
+/*! Decode a buffer of G.722 data to linear PCM.
+    \param s The G.722 context.
+    \param amp The audio sample buffer.
+    \param g722_data
+    \param len
+    \return The number of samples returned. */
+int g722_decode(g722_decode_state_t *s, int16_t amp[], const uint8_t g722_data[], int len);
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/g726.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/g726.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,195 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * g726.h - ITU G.726 codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * Based on G.721/G.723 code which is:
+ *
+ * This source code is a product of Sun Microsystems, Inc. and is provided
+ * for unrestricted use.  Users may copy or modify this source code without
+ * charge.
+ *
+ * SUN SOURCE CODE IS PROVIDED AS IS WITH NO WARRANTIES OF ANY KIND INCLUDING
+ * THE WARRANTIES OF DESIGN, MERCHANTIBILITY AND FITNESS FOR A PARTICULAR
+ * PURPOSE, OR ARISING FROM A COURSE OF DEALING, USAGE OR TRADE PRACTICE.
+ *
+ * Sun source code is provided with no support and without any obligation on
+ * the part of Sun Microsystems, Inc. to assist in its use, correction,
+ * modification or enhancement.
+ *
+ * SUN MICROSYSTEMS, INC. SHALL HAVE NO LIABILITY WITH RESPECT TO THE
+ * INFRINGEMENT OF COPYRIGHTS, TRADE SECRETS OR ANY PATENTS BY THIS SOFTWARE
+ * OR ANY PART THEREOF.
+ *
+ * In no event will Sun Microsystems, Inc. be liable for any lost revenue
+ * or profits or other special, indirect and consequential damages, even if
+ * Sun has been advised of the possibility of such damages.
+ *
+ * Sun Microsystems, Inc.
+ * 2550 Garcia Avenue
+ * Mountain View, California  94043
+ *
+ * $Id: g726.h,v 1.18 2007/12/13 11:31:32 steveu Exp $
+ */
+
+/*! \file */
+
+#if !defined(_SPANDSP_G726_H_)
+#define _SPANDSP_G726_H_
+
+/*! \page g726_page G.726 encoding and decoding
+\section g726_page_sec_1 What does it do?
+
+The G.726 module is a bit exact implementation of the full ITU G.726 specification.
+It supports:
+    - 16 kbps, 24kbps, 32kbps, and 40kbps operation.
+    - Tandem adjustment, for interworking with A-law and u-law.
+    - Annex A support, for use in environments not using A-law or u-law.
+
+It passes the ITU tests.
+
+\section g726_page_sec_2 How does it work?
+???.
+*/
+
+enum
+{
+    G726_ENCODING_LINEAR = 0,   /* Interworking with 16 bit signed linear */
+    G726_ENCODING_ULAW,         /* Interworking with u-law */
+    G726_ENCODING_ALAW          /* Interworking with A-law */
+};
+
+enum
+{
+    G726_PACKING_NONE = 0,
+    G726_PACKING_LEFT = 1,
+    G726_PACKING_RIGHT = 2
+};
+
+struct g726_state_s;
+
+typedef int16_t (*g726_decoder_func_t)(struct g726_state_s *s, uint8_t code);
+
+typedef uint8_t (*g726_encoder_func_t)(struct g726_state_s *s, int16_t amp);
+
+/*!
+ * The following is the definition of the state structure
+ * used by the G.726 encoder and decoder to preserve their internal
+ * state between successive calls.  The meanings of the majority
+ * of the state structure fields are explained in detail in the
+ * CCITT Recommendation G.721.  The field names are essentially indentical
+ * to variable names in the bit level description of the coding algorithm
+ * included in this Recommendation.
+ */
+typedef struct g726_state_s
+{
+    /*! The bit rate */
+    int rate;
+    /*! The external coding, for tandem operation */
+    int ext_coding;
+    /*! The number of bits per sample */
+    unsigned int bits_per_sample;
+    /*! One of the G.726_PACKING_xxx options */
+    int packing;
+
+    /*! Locked or steady state step size multiplier. */
+    int32_t yl;
+    /*! Unlocked or non-steady state step size multiplier. */
+    int16_t yu;
+    /*! int16_t term energy estimate. */
+    int16_t dms;
+    /*! Long term energy estimate. */
+    int16_t dml;
+    /*! Linear weighting coefficient of 'yl' and 'yu'. */
+    int16_t ap;
+    
+    /*! Coefficients of pole portion of prediction filter. */
+    int16_t a[2];
+    /*! Coefficients of zero portion of prediction filter. */
+    int16_t b[6];
+    /*! Signs of previous two samples of a partially reconstructed signal. */
+    int16_t pk[2];
+    /*! Previous 6 samples of the quantized difference signal represented in
+        an internal floating point format. */
+    int16_t dq[6];
+    /*! Previous 2 samples of the quantized difference signal represented in an
+        internal floating point format. */
+    int16_t sr[2];
+    /*! Delayed tone detect */
+    int td;
+    
+    /*! \brief The bit stream processing context. */
+    bitstream_state_t bs;
+
+    /*! \brief The current encoder function. */
+    g726_encoder_func_t enc_func;
+    /*! \brief The current decoder function. */
+    g726_decoder_func_t dec_func;
+} g726_state_t;
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+/*! Initialise a G.726 encode or decode context.
+    \param s The G.726 context.
+    \param bit_rate The required bit rate for the ADPCM data.
+           The valid rates are 16000, 24000, 32000 and 40000.
+    \param ext_coding The coding used outside G.726.
+    \param packing One of the G.726_PACKING_xxx options.
+    \return A pointer to the G.726 context, or NULL for error. */
+g726_state_t *g726_init(g726_state_t *s, int bit_rate, int ext_coding, int packing);
+
+/*! Free a G.726 encode or decode context.
+    \param s The G.726 context.
+    \return 0 for OK. */
+int g726_release(g726_state_t *s);
+
+/*! Decode a buffer of G.726 ADPCM data to linear PCM, a-law or u-law.
+    \param s The G.726 context.
+    \param amp The audio sample buffer.
+    \param g726_data
+    \param g726_bytes
+    \return The number of samples returned. */
+int g726_decode(g726_state_t *s,
+                int16_t amp[],
+                const uint8_t g726_data[],
+                int g726_bytes);
+
+/*! Encode a buffer of linear PCM data to G.726 ADPCM.
+    \param s The G.726 context.
+    \param g726_data The G.726 data produced.
+    \param amp The audio sample buffer.
+    \param len The number of samples in the buffer.
+    \return The number of bytes of G.726 data produced. */
+int g726_encode(g726_state_t *s,
+                uint8_t g726_data[],
+                const int16_t amp[],
+                int len);
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/gsm0610.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/gsm0610.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,175 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * gsm0610.h - GSM 06.10 full rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: gsm0610.h,v 1.13 2007/12/13 11:31:32 steveu Exp $
+ */
+
+#if !defined(_SPANDSP_GSM0610_H_)
+#define _SPANDSP_GSM0610_H_
+
+/*! \page gsm0610_page GSM 06.10 encoding and decoding
+\section gsm0610_page_sec_1 What does it do?
+
+The GSM 06.10 module is an version of the widely used GSM FR codec software
+available from http://kbs.cs.tu-berlin.de/~jutta/toast.html. This version
+was produced since some versions of this codec are not bit exact, or not
+very efficient on modern processors. This implementation can use MMX instructions
+on Pentium class processors, or alternative methods on other processors. It
+passes all the ETSI test vectors. That is, it is a tested bit exact implementation.
+
+This implementation supports encoded data in one of three packing formats:
+    - Unpacked, with the 76 parameters of a GSM 06.10 code frame each occupying a
+      separate byte. (note that none of the parameters exceed 8 bits).
+    - Packed the the 33 byte per frame, used for VoIP, where 4 bits per frame are wasted.
+    - Packed in WAV49 format, where 2 frames are packed into 65 bytes.
+
+\section gsm0610_page_sec_2 How does it work?
+???.
+*/
+
+enum
+{
+    GSM0610_PACKING_NONE,
+    GSM0610_PACKING_WAV49,
+    GSM0610_PACKING_VOIP
+};
+
+/*!
+    GSM 06.10 FR codec unpacked frame.
+*/
+typedef struct
+{
+    int16_t LARc[8];
+    int16_t Nc[4];
+    int16_t bc[4];
+    int16_t Mc[4];
+    int16_t xmaxc[4];
+    int16_t xMc[4][13];
+} gsm0610_frame_t;
+
+/*!
+    GSM 06.10 FR codec state descriptor. This defines the state of
+    a single working instance of the GSM 06.10 FR encoder or decoder.
+*/
+typedef struct
+{
+    /*! \brief One of the packing modes */
+    int packing;
+
+    int16_t dp0[280];
+
+    /*! Preprocessing */
+    int16_t z1;
+    int32_t L_z2;
+    /*! Pre-emphasis */
+    int16_t mp;
+
+    /*! Short term delay filter */
+    int16_t u[8];
+    int16_t LARpp[2][8];
+    int16_t j;
+
+    /*! Long term synthesis */
+    int16_t nrp;
+    /*! Short term synthesis */
+    int16_t v[9];
+    /*! Decoder postprocessing */
+    int16_t msr;
+    
+    /*! Encoder data */
+    int16_t e[50];
+} gsm0610_state_t;
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+/*! Initialise a GSM 06.10 encode or decode context.
+    \param s The GSM 06.10 context
+    \param packing One of the GSM0610_PACKING_xxx options.
+    \return A pointer to the GSM 06.10 context, or NULL for error. */
+gsm0610_state_t *gsm0610_init(gsm0610_state_t *s, int packing);
+
+/*! Release a GSM 06.10 encode or decode context.
+    \param s The GSM 06.10 context
+    \return 0 for success, else -1. */
+int gsm0610_release(gsm0610_state_t *s);
+
+/*! Set the packing format for a GSM 06.10 encode or decode context.
+    \param s The GSM 06.10 context
+    \param packing One of the GSM0610_PACKING_xxx options.
+    \return 0 for success, else -1. */
+int gsm0610_set_packing(gsm0610_state_t *s, int packing);
+
+/*! Encode a buffer of linear PCM data to GSM 06.10.
+    \param s The GSM 06.10 context.
+    \param code The GSM 06.10 data produced.
+    \param amp The audio sample buffer.
+    \param quant The number of samples in the buffer.
+    \return The number of bytes of GSM 06.10 data produced. */
+int gsm0610_encode(gsm0610_state_t *s, uint8_t code[], const int16_t amp[], int quant);
+
+/*! Decode a buffer of GSM 06.10 data to linear PCM.
+    \param s The GSM 06.10 context.
+    \param amp The audio sample buffer.
+    \param code The GSM 06.10 data.
+    \param quant The number of frames of GSM 06.10 data to be decoded.
+    \return The number of samples returned. */
+int gsm0610_decode(gsm0610_state_t *s, int16_t amp[], const uint8_t code[], int quant);
+
+int gsm0610_pack_none(uint8_t c[], const gsm0610_frame_t *s);
+
+/*! Pack a pair of GSM 06.10 frames in the format used for wave files (wave type 49).
+    \param c The buffer for the packed data. This must be at least 65 bytes long.
+    \param s A pointer to the frames to be packed.
+    \return The number of bytes generated. */
+int gsm0610_pack_wav49(uint8_t c[], const gsm0610_frame_t *s);
+
+/*! Pack a GSM 06.10 frames in the format used for VoIP.
+    \param c The buffer for the packed data. This must be at least 33 bytes long.
+    \param s A pointer to the frame to be packed.
+    \return The number of bytes generated. */
+int gsm0610_pack_voip(uint8_t c[], const gsm0610_frame_t *s);
+
+int gsm0610_unpack_none(gsm0610_frame_t *s, const uint8_t c[]);
+
+/*! Unpack a pair of GSM 06.10 frames from the format used for wave files (wave type 49).
+    \param s A pointer to a buffer into which the frames will be packed.
+    \param c The buffer containing the data to be unpacked. This must be at least 65 bytes long.
+    \return The number of bytes absorbed. */
+int gsm0610_unpack_wav49(gsm0610_frame_t *s, const uint8_t c[]);
+
+/*! Unpack a GSM 06.10 frame from the format used for VoIP.
+    \param s A pointer to a buffer into which the frame will be packed.
+    \param c The buffer containing the data to be unpacked. This must be at least 33 bytes long.
+    \return The number of bytes absorbed. */
+int gsm0610_unpack_voip(gsm0610_frame_t *s, const uint8_t c[]);
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+/*- End of include ---------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/ima_adpcm.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/ima_adpcm.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,113 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * imaadpcm.c - Conversion routines between linear 16 bit PCM data and
+ *		        IMA/DVI/Intel ADPCM format.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2004 Steve Underwood
+ *
+ * Based on a bit from here, a bit from there, eye of toad,
+ * ear of bat, etc - plus, of course, my own 2 cents.
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: ima_adpcm.h,v 1.16 2007/12/13 11:31:32 steveu Exp $
+ */
+
+/*! \file */
+
+#if !defined(_SPANDSP_IMA_ADPCM_H_)
+#define _SPANDSP_IMA_ADPCM_H_
+
+/*! \page ima_adpcm_page IMA/DVI/Intel ADPCM encoding and decoding
+\section ima_adpcm_page_sec_1 What does it do?
+IMA ADPCM offers a good balance of simplicity and quality at a rate of
+32kbps.
+
+\section ima_adpcm_page_sec_2 How does it work?
+
+\section ima_adpcm_page_sec_3 How do I use it?
+*/
+
+enum
+{
+    IMA_ADPCM_DVI4 = 0,
+    IMA_ADPCM_VDVI = 1
+};
+
+/*!
+    IMA (DVI/Intel) ADPCM conversion state descriptor. This defines the state of
+    a single working instance of the IMA ADPCM converter. This is used for
+    either linear to ADPCM or ADPCM to linear conversion.
+*/
+typedef struct
+{
+    int variant;
+    /*! \brief The last state of the ADPCM algorithm. */
+    int last;
+    /*! \brief Current index into the step size table. */
+    int step_index;
+    /*! \brief The current IMA code byte in progress. */
+    uint16_t ima_byte;
+    int bits;
+} ima_adpcm_state_t;
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+/*! Initialise an IMA ADPCM encode or decode context.
+    \param s The IMA ADPCM context
+    \param variant ???
+    \return A pointer to the IMA ADPCM context, or NULL for error. */
+ima_adpcm_state_t *ima_adpcm_init(ima_adpcm_state_t *s, int variant);
+
+/*! Free an IMA ADPCM encode or decode context.
+    \param s The IMA ADPCM context.
+    \return 0 for OK. */
+int ima_adpcm_release(ima_adpcm_state_t *s);
+
+/*! Encode a buffer of linear PCM data to IMA ADPCM.
+    \param s The IMA ADPCM context.
+    \param ima_data The IMA ADPCM data produced.
+    \param amp The audio sample buffer.
+    \param len The number of samples in the buffer.
+    \return The number of bytes of IMA ADPCM data produced. */
+int ima_adpcm_encode(ima_adpcm_state_t *s,
+                     uint8_t ima_data[],
+                     const int16_t amp[],
+                     int len);
+
+/*! Decode a buffer of IMA ADPCM data to linear PCM.
+    \param s The IMA ADPCM context.
+    \param amp The audio sample buffer.
+    \param ima_data
+    \param ima_bytes
+    \return The number of samples returned. */
+int ima_adpcm_decode(ima_adpcm_state_t *s,
+                     int16_t amp[],
+                     const uint8_t ima_data[],
+                     int ima_bytes);
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/lpc10.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/lpc10.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,213 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * lpc10.h - LPC10 low bit rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: lpc10.h,v 1.13 2007/04/08 08:16:18 steveu Exp $
+ */
+
+#if !defined(_SPANDSP_LPC10_H_)
+#define _SPANDSP_LPC10_H_
+
+/*! \page lpc10_page LPC10 encoding and decoding
+\section lpc10_page_sec_1 What does it do?
+The LPC10 module implements the US Department of Defense LPC10
+codec. This codec produces compressed data at 2400bps. At such
+a low rate high fidelity cannot be expected. However, the speech
+clarity is quite good, and this codec is unencumbered by patent
+or other restrictions.
+
+\section lpc10_page_sec_2 How does it work?
+???.
+*/
+
+#define LPC10_SAMPLES_PER_FRAME 180
+#define LPC10_BITS_IN_COMPRESSED_FRAME 54
+
+/*!
+    LPC10 codec unpacked frame.
+*/
+typedef struct
+{
+    int32_t ipitch;
+    int32_t irms;
+    int32_t irc[10];
+} lpc10_frame_t;
+
+/*!
+    LPC10 codec encoder state descriptor. This defines the state of
+    a single working instance of the LPC10 encoder.
+*/
+typedef struct
+{
+    int error_correction;
+
+    /* State used only by function high_pass_100hz */
+    float z11;
+    float z21;
+    float z12;
+    float z22;
+    
+    /* State used by function lpc10_analyse */
+    float inbuf[LPC10_SAMPLES_PER_FRAME*3];
+    float pebuf[LPC10_SAMPLES_PER_FRAME*3];
+    float lpbuf[696];
+    float ivbuf[312];
+    float bias;
+    int32_t osbuf[10];      /* No initial value necessary */
+    int32_t osptr;          /* Initial value 1 */
+    int32_t obound[3];
+    int32_t vwin[3][2];     /* Initial value vwin[2][0] = 307; vwin[2][1] = 462; */
+    int32_t awin[3][2];     /* Initial value awin[2][0] = 307; awin[2][1] = 462; */
+    int32_t voibuf[4][2];
+    float rmsbuf[3];
+    float rcbuf[3][10];
+    float zpre;
+
+    /* State used by function onset */
+    float n;
+    float d__;          /* Initial value 1.0f */
+    float fpc;          /* No initial value necessary */
+    float l2buf[16];
+    float l2sum1;
+    int32_t l2ptr1;     /* Initial value 1 */
+    int32_t l2ptr2;     /* Initial value 9 */
+    int32_t lasti;      /* No initial value necessary */
+    int hyst;           /* Initial value FALSE */
+
+    /* State used by function lpc10_voicing */
+    float dither;       /* Initial value 20.0f */
+    float snr;
+    float maxmin;
+    float voice[3][2];  /* Initial value is probably unnecessary */
+    int32_t lbve;
+    int32_t lbue;
+    int32_t fbve;
+    int32_t fbue;
+    int32_t ofbue;
+    int32_t sfbue;
+    int32_t olbue;
+    int32_t slbue;
+
+    /* State used by function dynamic_pitch_tracking */
+    float s[60];
+    int32_t p[2][60];
+    int32_t ipoint;
+    float alphax;
+
+    /* State used by function lpc10_pack */
+    int32_t isync;
+} lpc10_encode_state_t;
+
+/*!
+    LPC10 codec decoder state descriptor. This defines the state of
+    a single working instance of the LPC10 decoder.
+*/
+typedef struct
+{
+    int error_correction;
+
+    /* State used by function decode */
+    int32_t iptold;     /* Initial value 60 */
+    int first;          /* Initial value TRUE */
+    int32_t ivp2h;
+    int32_t iovoic;
+    int32_t iavgp;      /* Initial value 60 */
+    int32_t erate;
+    int32_t drc[10][3];
+    int32_t dpit[3];
+    int32_t drms[3];
+
+    /* State used by function synths */
+    float buf[LPC10_SAMPLES_PER_FRAME*2];
+    int32_t buflen;     /* Initial value LPC10_SAMPLES_PER_FRAME */
+
+    /* State used by function pitsyn */
+    int32_t ivoico;     /* No initial value necessary as long as first_pitsyn is initially TRUE_ */
+    int32_t ipito;      /* No initial value necessary as long as first_pitsyn is initially TRUE_ */
+    float rmso;         /* Initial value 1.0f */
+    float rco[10];      /* No initial value necessary as long as first_pitsyn is initially TRUE_ */
+    int32_t jsamp;      /* Nno initial value necessary as long as first_pitsyn is initially TRUE_ */
+    int first_pitsyn;   /* Initial value TRUE */
+
+    /* State used by function bsynz */
+    int32_t ipo;
+    float exc[166];
+    float exc2[166];
+    float lpi[3];
+    float hpi[3];
+    float rmso_bsynz;
+
+    /* State used by function random */
+    int32_t j;
+    int32_t k;
+    int16_t y[5];
+
+    /* State used by function deemp */
+    float dei[2];
+    float deo[3];
+} lpc10_decode_state_t;
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+/*! Initialise an LPC10e encode context.
+    \param s The LPC10e context
+    \param error_correction ???
+    \return A pointer to the LPC10e context, or NULL for error. */
+lpc10_encode_state_t *lpc10_encode_init(lpc10_encode_state_t *s, int error_correction);
+
+int lpc10_encode_release(lpc10_encode_state_t *s);
+
+/*! Encode a buffer of linear PCM data to LPC10e.
+    \param s The LPC10e context.
+    \param ima_data The LPC10e data produced.
+    \param amp The audio sample buffer.
+    \param len The number of samples in the buffer.
+    \return The number of bytes of LPC10e data produced. */
+int lpc10_encode(lpc10_encode_state_t *s, uint8_t code[], const int16_t amp[], int quant);
+
+/*! Initialise an LPC10e decode context.
+    \param s The LPC10e context
+    \param error_correction ???
+    \return A pointer to the LPC10e context, or NULL for error. */
+lpc10_decode_state_t *lpc10_decode_init(lpc10_decode_state_t *st, int error_correction);
+
+int lpc10_decode_release(lpc10_decode_state_t *s);
+
+/*! Decode a buffer of LPC10e data to linear PCM.
+    \param s The LPC10e context.
+    \param amp The audio sample buffer.
+    \param code The LPC10e data.
+    \param quant The number of frames of LPC10e data to be decoded.
+    \return The number of samples returned. */
+int lpc10_decode(lpc10_decode_state_t *s, int16_t amp[], const uint8_t code[], int quant);
+
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+/*- End of include ---------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/oki_adpcm.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/oki_adpcm.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,119 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * oki_adpcm.h - Conversion routines between linear 16 bit PCM data and
+ *		         OKI (Dialogic) ADPCM format.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2001 Steve Underwood
+ *
+ * Based on a bit from here, a bit from there, eye of toad,
+ * ear of bat, etc - plus, of course, my own 2 cents.
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: oki_adpcm.h,v 1.17 2007/12/13 11:31:33 steveu Exp $
+ */
+
+/*! \file */
+
+#if !defined(_SPANDSP_OKI_ADPCM_H_)
+#define _SPANDSP_OKI_ADPCM_H_
+
+/*! \page okiadpcm_page OKI (Dialogic) ADPCM encoding and decoding
+\section okiadpcm_page_sec_1 What does it do?
+OKI ADPCM is widely used in the CTI industry because it is the principal format
+supported by Dialogic. As the market leader, they tend to define "common
+practice". It offers a good balance of simplicity and quality at rates of
+24kbps or 32kbps. 32kbps is obtained by ADPCM compressing 8k samples/second linear
+PCM. 24kbps is obtained by resampling to 6k samples/second and using the same ADPCM
+compression algorithm on the slower samples.
+
+The algorithms for this ADPCM codec can be found in "PC Telephony - The complete guide
+to designing, building and programming systems using Dialogic and Related Hardware"
+by Bob Edgar. pg 272-276. */
+
+/*!
+    Oki (Dialogic) ADPCM conversion state descriptor. This defines the state of
+    a single working instance of the Oki ADPCM converter. This is used for
+    either linear to ADPCM or ADPCM to linear conversion.
+*/
+typedef struct
+{
+    /*! \brief The bit rate - 24000 or 32000. */
+    int bit_rate;
+    /*! \brief The last state of the ADPCM algorithm. */
+    int16_t last;
+    /*! \brief Current index into the step size table. */
+    int16_t step_index;
+    /*! \brief The compressed data byte in progress. */
+    uint8_t oki_byte;
+    /*! \brief The signal history for the sample rate converter. */
+    int16_t history[32];
+    /*! \brief Pointer into the history buffer. */
+    int ptr;
+    /*! \brief Odd/even sample counter. */
+    int mark;
+    /*! \brief Phase accumulator for the sample rate converter. */
+    int phase;
+} oki_adpcm_state_t;
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+/*! Initialise an Oki ADPCM encode or decode context.
+    \param s The Oki ADPCM context.
+    \param bit_rate The required bit rate for the ADPCM data.
+           The valid rates are 24000 and 32000.
+    \return A pointer to the Oki ADPCM context, or NULL for error. */
+oki_adpcm_state_t *oki_adpcm_init(oki_adpcm_state_t *s, int bit_rate);
+
+/*! Free an Oki ADPCM encode or decode context.
+    \param s The Oki ADPCM context.
+    \return 0 for OK. */
+int oki_adpcm_release(oki_adpcm_state_t *s);
+
+/*! Decode a buffer of Oki ADPCM data to linear PCM.
+    \param s The Oki ADPCM context.
+    \param amp The audio sample buffer.
+    \param oki_data
+    \param oki_bytes
+    \return The number of samples returned. */
+int oki_adpcm_decode(oki_adpcm_state_t *s,
+                     int16_t amp[],
+                     const uint8_t oki_data[],
+                     int oki_bytes);
+
+/*! Encode a buffer of linear PCM data to Oki ADPCM.
+    \param s The Oki ADPCM context.
+    \param oki_data The Oki ADPCM data produced
+    \param amp The audio sample buffer.
+    \param len The number of samples in the buffer.
+    \return The number of bytes of Oki ADPCM data produced. */
+int oki_adpcm_encode(oki_adpcm_state_t *s,
+                     uint8_t oki_data[],
+                     const int16_t amp[],
+                     int len);
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/telephony.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/telephony.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,67 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * telephony.h - some very basic telephony definitions
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2003 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: telephony.h,v 1.10 2007/04/05 19:20:50 steveu Exp $
+ */
+
+#if !defined(_SPANDSP_TELEPHONY_H_)
+#define _SPANDSP_TELEPHONY_H_
+
+#define SAMPLE_RATE             8000
+
+/* This is based on A-law, but u-law is only 0.03dB different */
+#define DBM0_MAX_POWER          (3.14f + 3.02f)
+#define DBM0_MAX_SINE_POWER     (3.14f)
+/* This is based on the ITU definition of dbOv in G.100.1 */
+#define DBOV_MAX_POWER          (0.0f)
+#define DBOV_MAX_SINE_POWER     (-3.02f)
+
+/*! \brief A handler for pure receive. The buffer cannot be altered. */
+typedef int (span_rx_handler_t)(void *s, const int16_t amp[], int len);
+
+/*! \brief A handler for receive, where the buffer can be altered. */
+typedef int (span_mod_handler_t)(void *s, int16_t amp[], int len);
+
+/*! \brief A handler for transmit, where the buffer will be filled. */
+typedef int (span_tx_handler_t)(void *s, int16_t amp[], int max_len);
+
+#define ms_to_samples(t)    (((t)*SAMPLE_RATE)/1000)
+
+#if !defined(FALSE)
+#define FALSE 0
+#endif
+#if !defined(TRUE)
+#define TRUE (!FALSE)
+#endif
+
+#if defined(__cplusplus)
+/* C++ doesn't seem to have sane rounding functions/macros yet */
+#ifndef _MSC_VER
+#define lrint(x) ((long int) (x))
+#define lrintf(x) ((long int) (x))
+#endif
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/vector_int.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/vector_int.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,99 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * vector_int.h
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2003 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the Lesser GNU General Public License version 2.1, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: vector_int.h,v 1.8 2007/12/13 11:31:33 steveu Exp $
+ */
+
+#if !defined(_SPANDSP_VECTOR_INT_H_)
+#define _SPANDSP_VECTOR_INT_H_
+
+#if defined(__cplusplus)
+extern "C"
+{
+#endif
+
+int32_t vec_dot_prodi16(const int16_t x[], const int16_t y[], int n);
+
+/*! \brief Find the minimum and maximum values in a vector.
+    \param x The vector to be searched.
+    \param n The number of elements in the vetor.
+    \param out A two element vector. The first will receive the 
+           maximum. The second will receive the minimum. This parameter
+           may be set to NULL.
+    \return The absolute maximum value. Since the range of negative numbers
+            exceeds the range of positive one, the returned integer is longer
+            than the ones being searched. */
+int32_t vec_min_maxi16(const int16_t x[], int n, int16_t out[]);
+
+static __inline__ int vec_norm2i16(const int16_t *vec, int len)
+{
+    int i;
+    int sum;
+
+    sum = 0;
+    for (i = 0;  i < len;  i++)
+        sum += vec[i]*vec[i];
+    return sum;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ void vec_sari16(int16_t *vec, int len, int shift)
+{
+    int i;
+
+    for (i = 0;  i < len;  i++)
+        vec[i] >>= shift;
+}
+/*- End of function --------------------------------------------------------*/
+
+static __inline__ int vec_max_bitsi16(const int16_t *vec, int len)
+{
+    int i;
+    int max;
+    int v;
+    int b;
+
+    max = 0;
+    for (i = 0;  i < len;  i++)
+    {
+        v = abs(vec[i]);
+        if (v > max)
+            max = v;
+    }
+    b = 0;
+    while (max != 0)
+    {
+        b++;
+        max >>= 1;
+    }
+    return b;
+}
+/*- End of function --------------------------------------------------------*/
+
+#if defined(__cplusplus)
+}
+#endif
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/version.h
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/version.h	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,37 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * version.h - A tag file, so the exact installed revision can be assertained.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2007 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: version.h.in,v 1.2 2007/04/06 13:20:36 steveu Exp $
+ */
+
+#if !defined(_SPANDSP_VERSION_H_)
+#define _SPANDSP_VERSION_H_
+
+/* The date and time of the version are in UTC form. */
+
+#define SPANDSP_RELEASE_DATE    $SPANDSP_RELEASE_DATE
+#define SPANDSP_RELEASE_TIME    $SPANDSP_RELEASE_TIME
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/src/voipcodecs/version.h.in
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/src/voipcodecs/version.h.in	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,37 @@
+/*
+ * SpanDSP - a series of DSP components for telephony
+ *
+ * version.h - A tag file, so the exact installed revision can be assertained.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2007 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: version.h.in,v 1.2 2007/04/06 13:20:36 steveu Exp $
+ */
+
+#if !defined(_SPANDSP_VERSION_H_)
+#define _SPANDSP_VERSION_H_
+
+/* The date and time of the version are in UTC form. */
+
+#define SPANDSP_RELEASE_DATE    $SPANDSP_RELEASE_DATE
+#define SPANDSP_RELEASE_TIME    $SPANDSP_RELEASE_TIME
+
+#endif
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/tests/Makefile.am
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/tests/Makefile.am	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,60 @@
+##
+## SpanDSP - a series of DSP components for telephony
+##
+## Makefile.am -- Process this file with automake to produce Makefile.in
+##
+## This program is free software; you can redistribute it and/or modify
+## it under the terms of the GNU General Public License version 2, as
+## published by the Free Software Foundation.
+##
+## This program is distributed in the hope that it will be useful,
+## but WITHOUT ANY WARRANTY; without even the implied warranty of
+## MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+## GNU General Public License for more details.
+##
+## You should have received a copy of the GNU General Public License
+## along with this program; if not, write to the Free Software
+## Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+##
+## $Id: Makefile.am,v 1.94 2008/01/31 13:34:40 steveu Exp $
+
+AM_CFLAGS = $(COMP_VENDOR_CFLAGS)
+
+LIBS += $(TESTLIBS)
+
+EXTRA_DIST = regression_tests.sh
+ 
+MAINTAINERCLEANFILES = Makefile.in
+
+INCLUDES = -I$(top_builddir)/src -DDATADIR="\"$(pkgdatadir)\""
+
+LIBDIR = -L$(top_builddir)/src
+
+noinst_PROGRAMS =   g711_tests \
+                    g722_tests \
+                    g726_tests \
+                    gsm0610_tests \
+                    ima_adpcm_tests \
+                    lpc10_tests \
+                    oki_adpcm_tests
+
+g711_tests_SOURCES = g711_tests.c
+g711_tests_LDADD = $(LIBDIR) -lvoipcodecs -lspandsp
+
+g722_tests_SOURCES = g722_tests.c
+g722_tests_LDADD = $(LIBDIR) -lvoipcodecs
+
+g726_tests_SOURCES = g726_tests.c
+g726_tests_LDADD = $(LIBDIR) -lvoipcodecs
+
+gsm0610_tests_SOURCES = gsm0610_tests.c
+gsm0610_tests_LDADD = $(LIBDIR) -lvoipcodecs
+
+ima_adpcm_tests_SOURCES = ima_adpcm_tests.c
+ima_adpcm_tests_LDADD = $(LIBDIR) -lvoipcodecs
+
+lpc10_tests_SOURCES = lpc10_tests.c
+lpc10_tests_LDADD = $(LIBDIR) -lvoipcodecs
+
+oki_adpcm_tests_SOURCES = oki_adpcm_tests.c
+oki_adpcm_tests_LDADD = $(LIBDIR) -lvoipcodecs

Added: freeswitch/trunk/libs/voipcodecs/tests/g711_tests.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/tests/g711_tests.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,287 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * g711_tests.c
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: g711_tests.c,v 1.7 2007/11/10 11:14:58 steveu Exp $
+ */
+
+/*! \page g711_tests_page A-law and u-law conversion tests
+\section g711_tests_page_sec_1 What does it do?
+
+\section g711_tests_page_sec_2 How is it used?
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <fcntl.h>
+#include <string.h>
+#include <audiofile.h>
+
+#include "voipcodecs.h"
+#include <spandsp/power_meter.h>
+
+#define OUT_FILE_NAME   "g711.wav"
+
+int16_t amp[65536];
+
+const uint8_t alaw_1khz_sine[] = {0x34, 0x21, 0x21, 0x34, 0xB4, 0xA1, 0xA1, 0xB4};
+const uint8_t ulaw_1khz_sine[] = {0x1E, 0x0B, 0x0B, 0x1E, 0x9E, 0x8B, 0x8B, 0x9E};
+
+int main(int argc, char *argv[])
+{
+    AFfilehandle outhandle;
+    AFfilesetup filesetup;
+    power_meter_t power_meter;
+    int outframes;
+    int i;
+    int block;
+    int pre;
+    int post;
+    int post_post;
+    int alaw_failures;
+    int ulaw_failures;
+    float worst_alaw;
+    float worst_ulaw;
+    float tmp;
+    
+    if ((filesetup = afNewFileSetup()) == AF_NULL_FILESETUP)
+    {
+        fprintf(stderr, "    Failed to create file setup\n");
+        exit(2);
+    }
+    afInitSampleFormat(filesetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+    afInitRate(filesetup, AF_DEFAULT_TRACK, (float) SAMPLE_RATE);
+    afInitFileFormat(filesetup, AF_FILE_WAVE);
+    afInitChannels(filesetup, AF_DEFAULT_TRACK, 1);
+
+    if ((outhandle = afOpenFile(OUT_FILE_NAME, "w", filesetup)) == AF_NULL_FILEHANDLE)
+    {
+        fprintf(stderr, "    Cannot create wave file '%s'\n", OUT_FILE_NAME);
+        exit(2);
+    }
+
+    printf("Conversion accuracy tests.\n");
+    alaw_failures = 0;
+    ulaw_failures = 0;
+    worst_alaw = 0.0;
+    worst_ulaw = 0.0;
+    for (block = 0;  block < 1;  block++)
+    {
+        for (i = 0;  i < 65536;  i++)
+        {
+            pre = i - 32768;
+            post = alaw_to_linear(linear_to_alaw(pre));
+            if (abs(pre) > 140)
+            {
+                tmp = (float) abs(post - pre)/(float) abs(pre);
+                if (tmp > 0.10)
+                {
+                    printf("A-law: Excessive error at %d (%d)\n", pre, post);
+                    alaw_failures++;
+                }
+                if (tmp > worst_alaw)
+                    worst_alaw = tmp;
+            }
+            else
+            {
+                /* Small values need different handling for sensible measurement */
+                if (abs(post - pre) > 15)
+                {
+                    printf("A-law: Excessive error at %d (%d)\n", pre, post);
+                    alaw_failures++;
+                }
+            }
+            amp[i] = post;
+        }
+        outframes = afWriteFrames(outhandle,
+                                  AF_DEFAULT_TRACK,
+                                  amp,
+                                  65536);
+        if (outframes != 65536)
+        {
+            fprintf(stderr, "    Error writing wave file\n");
+            exit(2);
+        }
+        for (i = 0;  i < 65536;  i++)
+        {
+            pre = i - 32768;
+            post = ulaw_to_linear(linear_to_ulaw(pre));
+            if (abs(pre) > 40)
+            {
+                tmp = (float) abs(post - pre)/(float) abs(pre);
+                if (tmp > 0.10)
+                {
+                    printf("u-law: Excessive error at %d (%d)\n", pre, post);
+                    ulaw_failures++;
+                }
+                if (tmp > worst_ulaw)
+                    worst_ulaw = tmp;
+            }
+            else
+            {
+                /* Small values need different handling for sensible measurement */
+                if (abs(post - pre) > 4)
+                {
+                    printf("u-law: Excessive error at %d (%d)\n", pre, post);
+                    ulaw_failures++;
+                }
+            }
+            amp[i] = post;
+        }
+        outframes = afWriteFrames(outhandle,
+                                  AF_DEFAULT_TRACK,
+                                  amp,
+                                  65536);
+        if (outframes != 65536)
+        {
+            fprintf(stderr, "    Error writing wave file\n");
+            exit(2);
+        }
+    }
+    printf("Worst A-law error (ignoring small values) %f%%\n", worst_alaw*100.0);
+    printf("Worst u-law error (ignoring small values) %f%%\n", worst_ulaw*100.0);
+    if (alaw_failures  ||  ulaw_failures)
+    {
+        printf("%d A-law values with excessive error\n", alaw_failures);
+        printf("%d u-law values with excessive error\n", ulaw_failures);
+        printf("Tests failed\n");
+        exit(2);
+    }
+    
+    printf("Cyclic conversion repeatability tests.\n");
+    /* Find what happens to every possible linear value after a round trip. */
+    for (i = 0;  i < 65536;  i++)
+    {
+        pre = i - 32768;
+        /* Make a round trip */
+        post = alaw_to_linear(linear_to_alaw(pre));
+        /* A second round trip should cause no further change */
+        post_post = alaw_to_linear(linear_to_alaw(post));
+        if (post_post != post)
+        {
+            printf("A-law second round trip mismatch - at %d, %d != %d\n", pre, post, post_post);
+            printf("Tests failed\n");
+            exit(2);
+        }
+        /* Make a round trip */
+        post = ulaw_to_linear(linear_to_ulaw(pre));
+        /* A second round trip should cause no further change */
+        post_post = ulaw_to_linear(linear_to_ulaw(post));
+        if (post_post != post)
+        {
+            printf("u-law round trip mismatch - at %d, %d != %d\n", pre, post, post_post);
+            printf("Tests failed\n");
+            exit(2);
+        }
+    }
+    
+    printf("Reference power level tests.\n");
+    power_meter_init(&power_meter, 7);
+
+    for (i = 0;  i < 8000;  i++)
+    {
+        amp[i] = ulaw_to_linear(ulaw_1khz_sine[i & 7]);
+        power_meter_update(&power_meter, amp[i]);
+    }
+    printf("Reference u-law 1kHz tone is %fdBm0\n", power_meter_current_dbm0(&power_meter));
+    outframes = afWriteFrames(outhandle,
+                              AF_DEFAULT_TRACK,
+                              amp,
+                              8000);
+    if (outframes != 8000)
+    {
+        fprintf(stderr, "    Error writing wave file\n");
+        exit(2);
+    }
+    if (0.1f < fabs(power_meter_current_dbm0(&power_meter)))
+    {
+        printf("Test failed.\n");
+        exit(2);
+    }
+
+    for (i = 0;  i < 8000;  i++)
+    {
+        amp[i] = alaw_to_linear(alaw_1khz_sine[i & 7]);
+        power_meter_update(&power_meter, amp[i]);
+    }
+    printf("Reference A-law 1kHz tone is %fdBm0\n", power_meter_current_dbm0(&power_meter));
+    outframes = afWriteFrames(outhandle,
+                              AF_DEFAULT_TRACK,
+                              amp,
+                              8000);
+    if (outframes != 8000)
+    {
+        fprintf(stderr, "    Error writing wave file\n");
+        exit(2);
+    }
+    if (0.1f < fabs(power_meter_current_dbm0(&power_meter)))
+    {
+        printf("Test failed.\n");
+        exit(2);
+    }
+
+    /* Check the transcoding functions. */
+    printf("Testing transcoding A-law -> u-law -> A-law\n");
+    for (i = 0;  i < 256;  i++)
+    {
+        if (alaw_to_ulaw(ulaw_to_alaw(i)) != i)
+        {
+            if (abs(alaw_to_ulaw(ulaw_to_alaw(i)) - i) > 1)
+            {
+                printf("u-law -> A-law -> u-law gave %d -> %d\n", i, alaw_to_ulaw(ulaw_to_alaw(i)));
+                printf("Test failed\n");
+                exit(2);
+            }
+        }
+    }
+
+    printf("Testing transcoding u-law -> A-law -> u-law\n");
+    for (i = 0;  i < 256;  i++)
+    {
+        if (ulaw_to_alaw(alaw_to_ulaw(i)) != i)
+        {
+            if (abs(alaw_to_ulaw(ulaw_to_alaw(i)) - i) > 1)
+            {
+                printf("A-law -> u-law -> A-law gave %d -> %d\n", i, ulaw_to_alaw(alaw_to_ulaw(i)));
+                printf("Test failed\n");
+                exit(2);
+            }
+        }
+    }
+    
+    if (afCloseFile(outhandle))
+    {
+        fprintf(stderr, "    Cannot close wave file '%s'\n", OUT_FILE_NAME);
+        exit(2);
+    }
+    afFreeFileSetup(filesetup);
+
+    printf("Tests passed.\n");
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/tests/g722_tests.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/tests/g722_tests.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,521 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * g722_tests.c - Test G.722 encode and decode.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2005 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: g722_tests.c,v 1.19 2007/11/10 11:14:58 steveu Exp $
+ */
+
+/*! \file */
+
+/*! \page g722_tests_page G.722 tests
+\section g722_tests_page_sec_1 What does it do?
+This modules implements two sets of tests:
+    - The tests defined in the G.722 specification, using the test data files supplied
+      with the specification.
+    - A generally audio quality test, consisting of compressing and decompressing a speeech
+      file for audible comparison.
+
+The speech file should be recorded at 16 bits/sample, 16000 samples/second, and named
+"pre_g722.wav".
+
+The ITU tests use the codec in a special mode, in which the QMFs, which split and recombine the
+sub-bands, are disabled. This means they do not test 100% of the codec. This is the reason for
+including the additional listening test.
+
+\section g722_tests_page_sec_2 How is it used?
+To perform the tests in the G.722 specification you need to obtain the test data files from the
+specification. These are copyright material, and so cannot be distributed with this test software.
+
+The files, containing test vectors, which are supplied with the G.722 specification, should be
+copied to itutests/g722. The ITU tests can then be run by executing g722_tests without
+any parameters.
+
+To perform a general audio quality test, g722_tests should be run with a parameter specifying
+the required bit rate for compression. The valid parameters are "-48", "-56", and "-64".
+The file ../localtests/short_wb_voice.wav will be compressed to the specified bit rate, decompressed,
+and the resulting audio stored in post_g722.wav.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <memory.h>
+#include <ctype.h>
+#include <audiofile.h>
+
+#include "voipcodecs.h"
+
+#define G722_SAMPLE_RATE    16000
+
+#define BLOCK_LEN           320
+
+#define MAX_TEST_VECTOR_LEN 40000
+
+#define TESTDATA_DIR        "../itutests/g722/"
+
+#define EIGHTK_IN_FILE_NAME "../localtests/short_nb_voice.wav"
+#define IN_FILE_NAME        "../localtests/short_wb_voice.wav"
+#define OUT_FILE_NAME       "post_g722.wav"
+
+#if 0
+static const char *itu_test_files[] =
+{
+    TESTDATA_DIR "T1C1.XMT",        /* 69973 bytes */
+    TESTDATA_DIR "T1C2.XMT",        /* 3605 bytes */
+    TESTDATA_DIR "T1D3.COD",        /* 69973 bytes */
+
+    TESTDATA_DIR "T2R1.COD",        /* 69973 bytes */
+    TESTDATA_DIR "T2R2.COD",        /* 3605 bytes */
+
+    TESTDATA_DIR "T3L1.RC1",        /* 69973 bytes */
+    TESTDATA_DIR "T3L1.RC2",        /* 69973 bytes */
+    TESTDATA_DIR "T3L1.RC3",        /* 69973 bytes */
+    TESTDATA_DIR "T3H1.RC0",        /* 69973 bytes */
+    TESTDATA_DIR "T3L2.RC1",        /* 3605 bytes */
+    TESTDATA_DIR "T3L2.RC2",        /* 3605 bytes */
+    TESTDATA_DIR "T3L2.RC3",        /* 3605 bytes */
+    TESTDATA_DIR "T3H2.RC0",        /* 3605 bytes */
+    TESTDATA_DIR "T3L3.RC1",        /* 69973 bytes */
+    TESTDATA_DIR "T3L3.RC2",        /* 69973 bytes */
+    TESTDATA_DIR "T3L3.RC3",        /* 69973 bytes */
+    TESTDATA_DIR "T3H3.RC0"         /* 69973 bytes */
+};
+#endif
+
+static const char *encode_test_files[] =
+{
+    TESTDATA_DIR "T1C1.XMT",    TESTDATA_DIR "T2R1.COD",
+    TESTDATA_DIR "T1C2.XMT",    TESTDATA_DIR "T2R2.COD",
+    NULL
+};
+
+static const char *decode_test_files[] =
+{
+    TESTDATA_DIR "T2R1.COD",
+    TESTDATA_DIR "T3L1.RC1",
+    TESTDATA_DIR "T3L1.RC2",
+    TESTDATA_DIR "T3L1.RC3",
+    TESTDATA_DIR "T3H1.RC0",
+
+    TESTDATA_DIR "T2R2.COD",
+    TESTDATA_DIR "T3L2.RC1",
+    TESTDATA_DIR "T3L2.RC2",
+    TESTDATA_DIR "T3L2.RC3",
+    TESTDATA_DIR "T3H2.RC0",
+
+    TESTDATA_DIR "T1D3.COD",
+    TESTDATA_DIR "T3L3.RC1",
+    TESTDATA_DIR "T3L3.RC2",
+    TESTDATA_DIR "T3L3.RC3",
+    TESTDATA_DIR "T3H3.RC0",
+    
+    NULL
+};
+
+int16_t itu_data[MAX_TEST_VECTOR_LEN];
+uint16_t itu_ref[MAX_TEST_VECTOR_LEN];
+uint16_t itu_ref_upper[MAX_TEST_VECTOR_LEN];
+uint8_t compressed[MAX_TEST_VECTOR_LEN];
+int16_t decompressed[MAX_TEST_VECTOR_LEN];
+
+static int hex_get(char *s)
+{
+    int i;
+    int value;
+    int x;
+
+    for (value = i = 0;  i < 4;  i++)
+    {
+        x = *s++ - 0x30;
+        if (x > 9)
+            x -= 0x07;
+        if (x > 15)
+            x -= 0x20;
+        if (x < 0  ||  x > 15)
+            return -1;
+        value <<= 4;
+        value |= x;
+    }
+    return value;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int get_vector(FILE *file, uint16_t vec[])
+{
+    char buf[132 + 1];
+    char *s;
+    int i;
+    int value;
+
+    while (fgets(buf, 133, file))
+    {
+        if (buf[0] == '/'  &&  buf[1] == '*')
+            continue;
+        s = buf;
+        i = 0;
+        while ((value = hex_get(s)) >= 0)
+        {
+            vec[i++] = value;
+            s += 4;
+        }
+        return i;
+    }
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int get_test_vector(const char *file, uint16_t buf[], int max_len)
+{
+    int octets;
+    int i;
+    FILE *infile;
+    
+    if ((infile = fopen(file, "r")) == NULL)
+    {
+        fprintf(stderr, "    Failed to open '%s'\n", file);
+        exit(2);
+    }
+    octets = 0;  
+    while ((i = get_vector(infile, buf + octets)) > 0)
+        octets += i;
+    fclose(infile);
+    return octets;
+}
+/*- End of function --------------------------------------------------------*/
+
+int main(int argc, char *argv[])
+{
+    g722_encode_state_t enc_state;
+    g722_decode_state_t dec_state;
+    int len;
+    int len_comp;
+    int len_comp_upper;
+    int len_data;
+    int len2;
+    int len3;
+    int i;
+    int j;
+    int k;
+    int file;
+    AFfilehandle inhandle;
+    AFfilehandle outhandle;
+    AFfilesetup filesetup;
+    int outframes;
+    int samples;
+    int mode;
+    int itutests;
+    int bit_rate;
+    int eight_k_in;
+    int eight_k_out;
+    float x;
+    int16_t indata[BLOCK_LEN];
+    int16_t outdata[BLOCK_LEN];
+    uint8_t adpcmdata[BLOCK_LEN];
+
+    i = 1;
+    bit_rate = 64000;
+    eight_k_in = FALSE;
+    eight_k_out = FALSE;
+    itutests = TRUE;
+    while (argc > i)
+    {
+        if (strcmp(argv[i], "-48") == 0)
+        {
+            bit_rate = 48000;
+            itutests = FALSE;
+            i++;
+        }
+        else if (strcmp(argv[i], "-56") == 0)
+        {
+            bit_rate = 56000;
+            itutests = FALSE;
+            i++;
+        }
+        else if (strcmp(argv[i], "-64") == 0)
+        {
+            bit_rate = 64000;
+            itutests = FALSE;
+            i++;
+        }
+        else if (strcmp(argv[i], "-8k8k") == 0)
+        {
+            eight_k_in = TRUE;
+            eight_k_out = TRUE;
+            i++;
+        }
+        else if (strcmp(argv[i], "-8k16k") == 0)
+        {
+            eight_k_in = TRUE;
+            eight_k_out = FALSE;
+            i++;
+        }
+        else if (strcmp(argv[i], "-16k8k") == 0)
+        {
+            eight_k_in = FALSE;
+            eight_k_out = TRUE;
+            i++;
+        }
+        else if (strcmp(argv[i], "-16k16k") == 0)
+        {
+            eight_k_in = FALSE;
+            eight_k_out = FALSE;
+            i++;
+        }
+        else
+        {
+            fprintf(stderr, "Unknown parameter %s specified.\n", argv[i]);
+            exit(2);
+        }
+    }
+
+    if (itutests)
+    {
+        /* ITU G.722 encode tests, using configuration 1. The QMF is bypassed */
+        for (file = 0;  encode_test_files[file];  file += 2)
+        {
+            printf("Testing %s -> %s\n", encode_test_files[file], encode_test_files[file + 1]);
+    
+            /* Get the input data */
+            len_data = get_test_vector(encode_test_files[file], (uint16_t *) itu_data, MAX_TEST_VECTOR_LEN);
+
+            /* Get the reference output data */
+            len_comp = get_test_vector(encode_test_files[file + 1], itu_ref, MAX_TEST_VECTOR_LEN);
+
+            /* Process the input data */
+            /* Skip the reset stuff at each end of the data */
+            for (i = 0;  i < len_data;  i++)
+            {
+                if ((itu_data[i] & 1) == 0)
+                    break;
+            }
+            for (j = i;  j < len_data;  j++)
+            {
+                if ((itu_data[j] & 1))
+                    break;
+            }
+            len = j - i;
+            g722_encode_init(&enc_state, 64000, 0);
+            enc_state.itu_test_mode = TRUE;
+            len2 = g722_encode(&enc_state, compressed, itu_data + i, len);
+
+            /* Check the result against the ITU's reference output data */
+            j = 0;
+            for (k = 0;  k < len2;  k++)
+            {
+                if ((compressed[k] & 0xFF) != ((itu_ref[k + i] >> 8) & 0xFF))
+                {
+                    printf(">>> %6d %4x %4x\n", k, compressed[k] & 0xFF, itu_ref[k + i] & 0xFFFF);
+                    j++;
+                }
+            }
+            printf("%d bad samples, out of %d/%d samples\n", j, len, len_data);
+            if (j)
+            {
+                printf("Test failed\n");
+                exit(2);
+            }
+            printf("Test passed\n");
+        }
+
+        /* ITU G.722 decode tests, using configuration 2. The QMF is bypassed */
+        /* Run each of the tests for each of the modes - 48kbps, 56kbps and 64kbps. */
+        for (mode = 1;  mode <= 3;  mode++)
+        {
+            for (file = 0;  decode_test_files[file];  file += 5)
+            {
+                printf("Testing mode %d, %s -> %s + %s\n",
+                       mode,
+                       decode_test_files[file],
+                       decode_test_files[file + mode],
+                       decode_test_files[file + 4]);
+
+                /* Get the input data */
+                len_data = get_test_vector(decode_test_files[file], (uint16_t *) itu_data, MAX_TEST_VECTOR_LEN);
+        
+                /* Get the lower reference output data */
+                len_comp = get_test_vector(decode_test_files[file + mode], itu_ref, MAX_TEST_VECTOR_LEN);
+        
+                /* Get the upper reference output data */
+                len_comp_upper = get_test_vector(decode_test_files[file + 4], itu_ref_upper, MAX_TEST_VECTOR_LEN);
+    
+                /* Process the input data */
+                /* Skip the reset stuff at each end of the data */
+                for (i = 0;  i < len_data;  i++)
+                {
+                    if ((itu_data[i] & 1) == 0)
+                        break;
+                }
+                for (j = i;  j < len_data;  j++)
+                {
+                    if ((itu_data[j] & 1))
+                        break;
+                }
+                len = j - i;
+                for (k = 0;  k < len;  k++)
+                    compressed[k] = itu_data[k + i] >> ((mode == 3)  ?  10  :  (mode == 2)  ?  9  :  8);
+        
+                g722_decode_init(&dec_state, (mode == 3)  ?  48000  :  (mode == 2)  ?  56000  :  64000, 0);
+                dec_state.itu_test_mode = TRUE;
+                len2 = g722_decode(&dec_state, decompressed, compressed, len);
+        
+                /* Check the result against the ITU's reference output data */
+                j = 0;
+                for (k = 0;  k < len2;  k += 2)
+                {
+                    if ((decompressed[k] & 0xFFFF) != (itu_ref[(k >> 1) + i] & 0xFFFF)
+                        ||
+                        (decompressed[k + 1] & 0xFFFF) != (itu_ref_upper[(k >> 1) + i] & 0xFFFF))
+                    {
+                        printf(">>> %6d %4x %4x %4x %4x\n", k >> 1, decompressed[k] & 0xFFFF, decompressed[k + 1] & 0xFFFF, itu_ref[(k >> 1) + i] & 0xFFFF, itu_ref_upper[(k >> 1) + i] & 0xFFFF);
+                        j++;
+                    }
+                }
+                printf("%d bad samples, out of %d/%d samples\n", j, len, len_data);
+                if (j)
+                {
+                    printf("Test failed\n");
+                    exit(2);
+                }
+                printf("Test passed\n");
+            }
+        }
+
+        printf("Tests passed.\n");
+    }
+    else
+    {
+        if (eight_k_in)
+        {
+            if ((inhandle = afOpenFile(EIGHTK_IN_FILE_NAME, "r", NULL)) == AF_NULL_FILEHANDLE)
+            {
+                fprintf(stderr, "    Cannot open wave file '%s'\n", EIGHTK_IN_FILE_NAME);
+                exit(2);
+            }
+            if ((x = afGetFrameSize(inhandle, AF_DEFAULT_TRACK, 1)) != 2.0)
+            {
+                printf("    Unexpected frame size in wave file '%s'\n", EIGHTK_IN_FILE_NAME);
+                exit(2);
+            }
+            if ((x = afGetRate(inhandle, AF_DEFAULT_TRACK)) != (float) SAMPLE_RATE)
+            {
+                printf("    Unexpected sample rate %f in wave file '%s'\n", x, EIGHTK_IN_FILE_NAME);
+                exit(2);
+            }
+            if ((x = afGetChannels(inhandle, AF_DEFAULT_TRACK)) != 1.0)
+            {
+                printf("    Unexpected number of channels in wave file '%s'\n", EIGHTK_IN_FILE_NAME);
+                exit(2);
+            }
+        }
+        else
+        {
+            if ((inhandle = afOpenFile(IN_FILE_NAME, "r", NULL)) == AF_NULL_FILEHANDLE)
+            {
+                fprintf(stderr, "    Cannot open wave file '%s'\n", IN_FILE_NAME);
+                exit(2);
+            }
+            if ((x = afGetFrameSize(inhandle, AF_DEFAULT_TRACK, 1)) != 2.0)
+            {
+                printf("    Unexpected frame size in wave file '%s'\n", IN_FILE_NAME);
+                exit(2);
+            }
+            if ((x = afGetRate(inhandle, AF_DEFAULT_TRACK)) != (float) G722_SAMPLE_RATE)
+            {
+                printf("    Unexpected sample rate %f in wave file '%s'\n", x, IN_FILE_NAME);
+                exit(2);
+            }
+            if ((x = afGetChannels(inhandle, AF_DEFAULT_TRACK)) != 1.0)
+            {
+                printf("    Unexpected number of channels in wave file '%s'\n", IN_FILE_NAME);
+                exit(2);
+            }
+        }
+        
+        if ((filesetup = afNewFileSetup()) == AF_NULL_FILESETUP)
+        {
+            fprintf(stderr, "    Failed to create file setup\n");
+            exit(2);
+        }
+        afInitSampleFormat(filesetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+        if (eight_k_out)
+            afInitRate(filesetup, AF_DEFAULT_TRACK, (float) SAMPLE_RATE);
+        else
+            afInitRate(filesetup, AF_DEFAULT_TRACK, (float) G722_SAMPLE_RATE);
+        afInitFileFormat(filesetup, AF_FILE_WAVE);
+        afInitChannels(filesetup, AF_DEFAULT_TRACK, 1);
+        if ((outhandle = afOpenFile(OUT_FILE_NAME, "w", filesetup)) == AF_NULL_FILEHANDLE)
+        {
+            fprintf(stderr, "    Cannot create wave file '%s'\n", OUT_FILE_NAME);
+            exit(2);
+        }
+        if (eight_k_in)
+            g722_encode_init(&enc_state, bit_rate, G722_PACKED | G722_SAMPLE_RATE_8000);
+        else
+            g722_encode_init(&enc_state, bit_rate, G722_PACKED);
+        if (eight_k_out)
+            g722_decode_init(&dec_state, bit_rate, G722_PACKED | G722_SAMPLE_RATE_8000);
+        else
+            g722_decode_init(&dec_state, bit_rate, G722_PACKED);
+        for (;;)
+        {
+            samples = afReadFrames(inhandle,
+                                   AF_DEFAULT_TRACK,
+                                   indata,
+                                   BLOCK_LEN);
+            if (samples <= 0)
+                break;
+            len2 = g722_encode(&enc_state, adpcmdata, indata, samples);
+            len3 = g722_decode(&dec_state, outdata, adpcmdata, len2);
+            outframes = afWriteFrames(outhandle,
+                                      AF_DEFAULT_TRACK,
+                                      outdata,
+                                      len3);
+            if (outframes != len3)
+            {
+                fprintf(stderr, "    Error writing wave file\n");
+                exit(2);
+            }
+        }
+        if (afCloseFile(inhandle))
+        {
+            fprintf(stderr, "    Cannot close wave file '%s'\n", IN_FILE_NAME);
+            exit(2);
+        }
+        if (afCloseFile(outhandle))
+        {
+            fprintf(stderr, "    Cannot close wave file '%s'\n", OUT_FILE_NAME);
+            exit(2);
+        }
+        afFreeFileSetup(filesetup);
+
+        printf("'%s' transcoded to '%s' at %dbps.\n", IN_FILE_NAME, OUT_FILE_NAME, bit_rate);
+    }
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/tests/g726_tests.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/tests/g726_tests.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,1334 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * g726_tests.c - Test G.726 encode and decode.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: g726_tests.c,v 1.17 2007/11/10 11:14:58 steveu Exp $
+ */
+
+/*! \file */
+
+/*! \page g726_tests_page G.726 tests
+\section g726_tests_page_sec_1 What does it do?
+Two sets of tests are performed:
+    - The tests defined in the G.726 specification, using the test data files supplied with
+      the specification.
+    - A generally audio quality test, consisting of compressing and decompressing a speeech
+      file for audible comparison.
+
+The speech file should be recorded at 16 bits/sample, 8000 samples/second, and named
+"pre_g726.wav".
+
+\section g726_tests_page_sec_2 How is it used?
+To perform the tests in the G.726 specification you need to obtain the test data files from the
+specification. These are copyright material, and so cannot be distributed with this test software.
+
+The files, containing test vectors, which are supplied with the G.726 specification, should be
+copied to itutests/g726 so the files are arranged in the same directory heirarchy in which they
+are supplied. That is, you should have file names like
+
+    - itutests/g726/DISK1/INPUT/NRM.M
+    - itutests/g726/DISK1/INPUT/OVR.M
+    - itutests/g726/DISK2/INPUT/NRM.A
+    - itutests/g726/DISK2/INPUT/OVR.A
+
+in your source tree. The ITU tests can then be run by executing g726_tests without
+any parameters.
+
+To perform a general audio quality test, g726_tests should be run with a parameter specifying
+the required bit rate for compression. The valid parameters are "-16", "-24", "-32", and "-40".
+The test file ../localtests/short_nb_voice.wav will be compressed to the specified bit rate,
+decompressed, and the resulting audio stored in post_g726.wav.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <memory.h>
+#include <ctype.h>
+#include <audiofile.h>
+
+#include "voipcodecs.h"
+
+#define BLOCK_LEN           320
+#define MAX_TEST_VECTOR_LEN 40000
+
+#define IN_FILE_NAME    "../localtests/short_nb_voice.wav"
+#define OUT_FILE_NAME   "post_g726.wav"
+
+#define TESTDATA_DIR    "../itutests/g726/"
+
+int16_t outdata[MAX_TEST_VECTOR_LEN];
+uint8_t adpcmdata[MAX_TEST_VECTOR_LEN];
+
+int16_t itudata[MAX_TEST_VECTOR_LEN];
+uint8_t itu_ref[MAX_TEST_VECTOR_LEN];
+uint8_t unpacked[MAX_TEST_VECTOR_LEN];
+uint8_t xlaw[MAX_TEST_VECTOR_LEN];
+
+/*
+Table 4 - V Reset and homing sequences for u-law
+            Normal                              I-input     Overload
+Algorithm   Input   Intermediate    Output      Input       Output      Input   Intermediate    Output
+            (PCM)   (ADPCM)         (PCM)       (ADPCM)     (PCM)       (PCM)   (ADPCM)         (PCM)
+
+16F         NRM.M   RN16FM.I        RN16FM.O    I16         RI16FM.O    OVR.M   RV16FM.I        RV16FM.O
+                    HN16FM.I        HN16FM.O                HI16FM.O            HV16FM.I        HV16FM.O
+
+24F         NRM.M   RN24FM.I        RN24FM.O    I24         RI24FM.O    OVR.M   RV24FM.I        RV24FM.O
+                    HN24FM.I        HN24FM.O                HI24FM.O            HV24FM.I        HV24FM.O
+
+32F         NRM.M   RN32FM.I        RN32FM.O    I32         RI32FM.O    OVR.M   RV32FM.I        RV32FM.O
+                    HN32FM.I        HN32FM.O                HI32FM.O            HV32FM.I        HV32FM.O
+
+40F         NRM.M   RN40FM.I        RN40FM.O    I40         RI40FM.O    OVR.M   RV40FM.I        RV40FM.O
+                    HN40FM.I        HN40FM.O                HI40FM.O            HV40FM.I        HV40FM.O
+
+
+Table 5 - V Reset and homing sequences for A-law
+            Normal                              I-input     Overload
+Algorithm   Input   Intermediate    Output      Input       Output      Input   Intermediate    Output
+            (PCM)   (ADPCM)         (PCM)       (ADPCM)     (PCM)       (PCM)   (ADPCM)         (PCM)
+16F         NRM.A   RN16FA.I        RN16FA.O    I16         RI16FA.O    OVR.A   RV16FA.I        RV16FA.O
+                    HN16FA.I        HN16FA.O                HI16FA.O            HV16FA.I        HV16FA.O
+
+24F         NRM.A   RN24FA.I        RN24FA.O    I24         RI24FA.O    OVR.A   RV24FA.I        RV24FA.O
+                    HN24FA.I        HN24FA.O                HI24FA.O            HV24FA.I        HV24FA.O
+
+32F         NRM.A   RN32FA.I        RN32FA.O    I32         RI32FA.O    OVR.A   RV32FA.I        RV32FA.O
+                    HN32FA.I        HN32FA.O                HI32FA.O            HV32FA.I        HV32FA.O
+
+40F         NRM.A   RN40FA.I        RN40FA.O    I40         RI40FA.O    OVR.A   RV40FA.I        RV40FA.O
+                    HN40FA.I        HN40FA.O                HI40FA.O            HV40FA.I        HV40FA.O
+
+Table 6 �V Reset and homing cross sequences for u-law -> A-law
+            Normal                              Overload
+Algorithm   Input   Intermediate    Output      Input   Intermediate    Output
+            (PCM)   (ADPCM)         (PCM)       (PCM)   (ADPCM)         (PCM)
+16F         NRM.M   RN16FM.I        RN16FC.O    OVR.M   RV16FM.I        RV16FC.O
+                    HN16FM.I        HN16FC.O            HV16FM.I        HV16FC.O
+
+24F         NRM.M   RN24FM.I        RN24FC.O    OVR.M   RV24FM.I        RV24FC.O
+                    HN24FM.I        HN24FC.O            HV24FM.I        HV24FC.O
+
+32F         NRM.M   RN32FM.I        RN32FC.O    OVR.M   RV32FM.I        RV32FC.O
+                    HN32FM.I        HN32FC.O            HV32FM.I        HV32FC.O
+
+40F         NRM.M   RN40FM.I        RN40FC.O    OVR.M   RV40FM.I        RV40FC.O
+                    HN40FM.I        HN40FC.O            HV40FM.I        HV40FC.O
+
+Table 7 �V Reset and homing cross sequences for A-law -> u-law
+            Normal                              Overload
+Algorithm   Input   Intermediate    Output      Input   Intermediate    Output
+            (PCM)   (ADPCM)         (PCM)       (PCM)   (ADPCM)         (PCM)
+16F         NRM.A   RN16FA.I        RN16FX.O    OVR.A   RV16FA.I        RV16FX.O
+                    HN16FA.I        HN16FX.O            HV16FA.I        HV16FX.O
+
+24F         NRM.A   RN24FA.I        RN24FX.O    OVR.A   RV24FA.I        RV24FX.O
+                    HN24FA.I        HN24FX.O            HV24FA.I        HV24FX.O
+
+32F         NRM.A   RN32FA.I        RN32FX.O    OVR.A   RV32FA.I        RV32FX.O
+                    HN32FA.I        HN32FX.O            HV32FA.I        HV32FX.O
+
+40F         NRM.A   RN40FA.I        RN40FX.O    OVR.A   RV40FA.I        RV40FX.O
+                    HN40FA.I        HN40FX.O            HV40FA.I        HV40FX.O
+*/
+
+#define	G726_ENCODING_NONE          9999
+
+typedef struct
+{
+    const char *conditioning_pcm_file;
+    const char *pcm_file;
+    const char *conditioning_adpcm_file;
+    const char *adpcm_file;
+    const char *output_file;
+    int rate;
+    int compression_law;
+    int decompression_law;
+} test_set_t;
+
+static test_set_t itu_test_sets[] =
+{
+    /* u-law to u-law tests */
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/16/RN16FM.I",
+        TESTDATA_DIR "DISK1/RESET/16/RN16FM.O",
+        16000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        "",
+        "",
+        TESTDATA_DIR "DISK1/INPUT/I16",
+        TESTDATA_DIR "DISK1/RESET/16/RI16FM.O",
+        16000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/16/RV16FM.I",
+        TESTDATA_DIR "DISK1/RESET/16/RV16FM.O",
+        16000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/24/RN24FM.I",
+        TESTDATA_DIR "DISK1/RESET/24/RN24FM.O",
+        24000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        "",
+        "",
+        TESTDATA_DIR "DISK1/INPUT/I24",
+        TESTDATA_DIR "DISK1/RESET/24/RI24FM.O",
+        24000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/24/RV24FM.I",
+        TESTDATA_DIR "DISK1/RESET/24/RV24FM.O",
+        24000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/32/RN32FM.I",
+        TESTDATA_DIR "DISK1/RESET/32/RN32FM.O",
+        32000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        "",
+        "",
+        TESTDATA_DIR "DISK1/INPUT/I32",
+        TESTDATA_DIR "DISK1/RESET/32/RI32FM.O",
+        32000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/32/RV32FM.I",
+        TESTDATA_DIR "DISK1/RESET/32/RV32FM.O",
+        32000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/40/RN40FM.I",
+        TESTDATA_DIR "DISK1/RESET/40/RN40FM.O",
+        40000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        "",
+        "",
+        TESTDATA_DIR "DISK1/INPUT/I40",
+        TESTDATA_DIR "DISK1/RESET/40/RI40FM.O",
+        40000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/40/RV40FM.I",
+        TESTDATA_DIR "DISK1/RESET/40/RV40FM.O",
+        40000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    /* A-law to A-law tests */
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/16/RN16FA.I",
+        TESTDATA_DIR "DISK2/RESET/16/RN16FA.O",
+        16000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        "",
+        "",
+        TESTDATA_DIR "DISK2/INPUT/I16",
+        TESTDATA_DIR "DISK2/RESET/16/RI16FA.O",
+        16000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/16/RV16FA.I",
+        TESTDATA_DIR "DISK2/RESET/16/RV16FA.O",
+        16000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/24/RN24FA.I",
+        TESTDATA_DIR "DISK2/RESET/24/RN24FA.O",
+        24000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        "",
+        "",
+        TESTDATA_DIR "DISK2/INPUT/I24",
+        TESTDATA_DIR "DISK2/RESET/24/RI24FA.O",
+        24000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/24/RV24FA.I",
+        TESTDATA_DIR "DISK2/RESET/24/RV24FA.O",
+        24000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/32/RN32FA.I",
+        TESTDATA_DIR "DISK2/RESET/32/RN32FA.O",
+        32000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        "",
+        "",
+        TESTDATA_DIR "DISK2/INPUT/I32",
+        TESTDATA_DIR "DISK2/RESET/32/RI32FA.O",
+        32000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/32/RV32FA.I",
+        TESTDATA_DIR "DISK2/RESET/32/RV32FA.O",
+        32000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/40/RN40FA.I",
+        TESTDATA_DIR "DISK2/RESET/40/RN40FA.O",
+        40000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        "",
+        "",
+        TESTDATA_DIR "DISK2/INPUT/I40",
+        TESTDATA_DIR "DISK2/RESET/40/RI40FA.O",
+        40000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/40/RV40FA.I",
+        TESTDATA_DIR "DISK2/RESET/40/RV40FA.O",
+        40000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    /* u-law to A-law tests */
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/16/RN16FM.I",
+        TESTDATA_DIR "DISK1/RESET/16/RN16FC.O",
+        16000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/16/RV16FM.I",
+        TESTDATA_DIR "DISK1/RESET/16/RV16FC.O",
+        16000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/24/RN24FM.I",
+        TESTDATA_DIR "DISK1/RESET/24/RN24FC.O",
+        24000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/24/RV24FM.I",
+        TESTDATA_DIR "DISK1/RESET/24/RV24FC.O",
+        24000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/32/RN32FM.I",
+        TESTDATA_DIR "DISK1/RESET/32/RN32FC.O",
+        32000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/32/RV32FM.I",
+        TESTDATA_DIR "DISK1/RESET/32/RV32FC.O",
+        32000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/40/RN40FM.I",
+        TESTDATA_DIR "DISK1/RESET/40/RN40FC.O",
+        40000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        "",
+        TESTDATA_DIR "DISK1/RESET/40/RV40FM.I",
+        TESTDATA_DIR "DISK1/RESET/40/RV40FC.O",
+        40000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    /* A-law to u-law tests */
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/16/RN16FA.I",
+        TESTDATA_DIR "DISK2/RESET/16/RN16FX.O",
+        16000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/16/RV16FA.I",
+        TESTDATA_DIR "DISK2/RESET/16/RV16FX.O",
+        16000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/24/RN24FA.I",
+        TESTDATA_DIR "DISK2/RESET/24/RN24FX.O",
+        24000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/24/RV24FA.I",
+        TESTDATA_DIR "DISK2/RESET/24/RV24FX.O",
+        24000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/32/RN32FA.I",
+        TESTDATA_DIR "DISK2/RESET/32/RN32FX.O",
+        32000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/32/RV32FA.I",
+        TESTDATA_DIR "DISK2/RESET/32/RV32FX.O",
+        32000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/40/RN40FA.I",
+        TESTDATA_DIR "DISK2/RESET/40/RN40FX.O",
+        40000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        "",
+        TESTDATA_DIR "DISK2/RESET/40/RV40FA.I",
+        TESTDATA_DIR "DISK2/RESET/40/RV40FX.O",
+        40000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    /* u-law to u-law tests */
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        TESTDATA_DIR "DISK1/HOMING/16/I_INI_16.M",
+        TESTDATA_DIR "DISK1/HOMING/16/HN16FM.I",
+        TESTDATA_DIR "DISK1/HOMING/16/HN16FM.O",
+        16000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        "",
+        TESTDATA_DIR "DISK1/HOMING/16/I_INI_16.M",
+        TESTDATA_DIR "DISK1/INPUT/I16",
+        TESTDATA_DIR "DISK1/HOMING/16/HI16FM.O",
+        16000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        TESTDATA_DIR "DISK1/HOMING/16/I_INI_16.M",
+        TESTDATA_DIR "DISK1/HOMING/16/HV16FM.I",
+        TESTDATA_DIR "DISK1/HOMING/16/HV16FM.O",
+        16000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        TESTDATA_DIR "DISK1/HOMING/24/I_INI_24.M",
+        TESTDATA_DIR "DISK1/HOMING/24/HN24FM.I",
+        TESTDATA_DIR "DISK1/HOMING/24/HN24FM.O",
+        24000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        "",
+        TESTDATA_DIR "DISK1/HOMING/24/I_INI_24.M",
+        TESTDATA_DIR "DISK1/INPUT/I24",
+        TESTDATA_DIR "DISK1/HOMING/24/HI24FM.O",
+        24000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        TESTDATA_DIR "DISK1/HOMING/24/I_INI_24.M",
+        TESTDATA_DIR "DISK1/HOMING/24/HV24FM.I",
+        TESTDATA_DIR "DISK1/HOMING/24/HV24FM.O",
+        24000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        TESTDATA_DIR "DISK1/HOMING/32/I_INI_32.M",
+        TESTDATA_DIR "DISK1/HOMING/32/HN32FM.I",
+        TESTDATA_DIR "DISK1/HOMING/32/HN32FM.O",
+        32000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        "",
+        TESTDATA_DIR "DISK1/HOMING/32/I_INI_32.M",
+        TESTDATA_DIR "DISK1/INPUT/I32",
+        TESTDATA_DIR "DISK1/HOMING/32/HI32FM.O",
+        32000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        TESTDATA_DIR "DISK1/HOMING/32/I_INI_32.M",
+        TESTDATA_DIR "DISK1/HOMING/32/HV32FM.I",
+        TESTDATA_DIR "DISK1/HOMING/32/HV32FM.O",
+        32000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        TESTDATA_DIR "DISK1/HOMING/40/I_INI_40.M",
+        TESTDATA_DIR "DISK1/HOMING/40/HN40FM.I",
+        TESTDATA_DIR "DISK1/HOMING/40/HN40FM.O",
+        40000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        "",
+        "",
+        TESTDATA_DIR "DISK1/HOMING/40/I_INI_40.M",
+        TESTDATA_DIR "DISK1/INPUT/I40",
+        TESTDATA_DIR "DISK1/HOMING/40/HI40FM.O",
+        40000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        TESTDATA_DIR "DISK1/HOMING/40/I_INI_40.M",
+        TESTDATA_DIR "DISK1/HOMING/40/HV40FM.I",
+        TESTDATA_DIR "DISK1/HOMING/40/HV40FM.O",
+        40000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ULAW
+    },
+    /* A-law to A-law tests */
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        TESTDATA_DIR "DISK2/HOMING/16/I_INI_16.A",
+        TESTDATA_DIR "DISK2/HOMING/16/HN16FA.I",
+        TESTDATA_DIR "DISK2/HOMING/16/HN16FA.O",
+        16000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        "",
+        TESTDATA_DIR "DISK2/HOMING/16/I_INI_16.A",
+        TESTDATA_DIR "DISK2/INPUT/I16",
+        TESTDATA_DIR "DISK2/HOMING/16/HI16FA.O",
+        16000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        TESTDATA_DIR "DISK2/HOMING/16/I_INI_16.A",
+        TESTDATA_DIR "DISK2/HOMING/16/HV16FA.I",
+        TESTDATA_DIR "DISK2/HOMING/16/HV16FA.O",
+        16000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        TESTDATA_DIR "DISK2/HOMING/24/I_INI_24.A",
+        TESTDATA_DIR "DISK2/HOMING/24/HN24FA.I",
+        TESTDATA_DIR "DISK2/HOMING/24/HN24FA.O",
+        24000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        "",
+        TESTDATA_DIR "DISK2/HOMING/24/I_INI_24.A",
+        TESTDATA_DIR "DISK2/INPUT/I24",
+        TESTDATA_DIR "DISK2/HOMING/24/HI24FA.O",
+        24000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        TESTDATA_DIR "DISK2/HOMING/24/I_INI_24.A",
+        TESTDATA_DIR "DISK2/HOMING/24/HV24FA.I",
+        TESTDATA_DIR "DISK2/HOMING/24/HV24FA.O",
+        24000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        TESTDATA_DIR "DISK2/HOMING/32/I_INI_32.A",
+        TESTDATA_DIR "DISK2/HOMING/32/HN32FA.I",
+        TESTDATA_DIR "DISK2/HOMING/32/HN32FA.O",
+        32000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        "",
+        TESTDATA_DIR "DISK2/HOMING/32/I_INI_32.A",
+        TESTDATA_DIR "DISK2/INPUT/I32",
+        TESTDATA_DIR "DISK2/HOMING/32/HI32FA.O",
+        32000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        TESTDATA_DIR "DISK2/HOMING/32/I_INI_32.A",
+        TESTDATA_DIR "DISK2/HOMING/32/HV32FA.I",
+        TESTDATA_DIR "DISK2/HOMING/32/HV32FA.O",
+        32000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        TESTDATA_DIR "DISK2/HOMING/40/I_INI_40.A",
+        TESTDATA_DIR "DISK2/HOMING/40/HN40FA.I",
+        TESTDATA_DIR "DISK2/HOMING/40/HN40FA.O",
+        40000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        "",
+        "",
+        TESTDATA_DIR "DISK2/HOMING/40/I_INI_40.A",
+        TESTDATA_DIR "DISK2/INPUT/I40",
+        TESTDATA_DIR "DISK2/HOMING/40/HI40FA.O",
+        40000,
+        G726_ENCODING_NONE,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        TESTDATA_DIR "DISK2/HOMING/40/I_INI_40.A",
+        TESTDATA_DIR "DISK2/HOMING/40/HV40FA.I",
+        TESTDATA_DIR "DISK2/HOMING/40/HV40FA.O",
+        40000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ALAW
+    },
+    /* u-law to A-law tests */
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        TESTDATA_DIR "DISK2/HOMING/16/I_INI_16.A",
+        TESTDATA_DIR "DISK1/HOMING/16/HN16FM.I",
+        TESTDATA_DIR "DISK1/HOMING/16/HN16FC.O",
+        16000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        TESTDATA_DIR "DISK2/HOMING/16/I_INI_16.A",
+        TESTDATA_DIR "DISK1/HOMING/16/HV16FM.I",
+        TESTDATA_DIR "DISK1/HOMING/16/HV16FC.O",
+        16000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        TESTDATA_DIR "DISK2/HOMING/24/I_INI_24.A",
+        TESTDATA_DIR "DISK1/HOMING/24/HN24FM.I",
+        TESTDATA_DIR "DISK1/HOMING/24/HN24FC.O",
+        24000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        TESTDATA_DIR "DISK2/HOMING/24/I_INI_24.A",
+        TESTDATA_DIR "DISK1/HOMING/24/HV24FM.I",
+        TESTDATA_DIR "DISK1/HOMING/24/HV24FC.O",
+        24000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        TESTDATA_DIR "DISK2/HOMING/32/I_INI_32.A",
+        TESTDATA_DIR "DISK1/HOMING/32/HN32FM.I",
+        TESTDATA_DIR "DISK1/HOMING/32/HN32FC.O",
+        32000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        TESTDATA_DIR "DISK2/HOMING/32/I_INI_32.A",
+        TESTDATA_DIR "DISK1/HOMING/32/HV32FM.I",
+        TESTDATA_DIR "DISK1/HOMING/32/HV32FC.O",
+        32000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/NRM.M",
+        TESTDATA_DIR "DISK2/HOMING/40/I_INI_40.A",
+        TESTDATA_DIR "DISK1/HOMING/40/HN40FM.I",
+        TESTDATA_DIR "DISK1/HOMING/40/HN40FC.O",
+        40000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    {
+        TESTDATA_DIR "DISK1/PCM_INIT.M",
+        TESTDATA_DIR "DISK1/INPUT/OVR.M",
+        TESTDATA_DIR "DISK2/HOMING/40/I_INI_40.A",
+        TESTDATA_DIR "DISK1/HOMING/40/HV40FM.I",
+        TESTDATA_DIR "DISK1/HOMING/40/HV40FC.O",
+        40000,
+        G726_ENCODING_ULAW,
+        G726_ENCODING_ALAW
+    },
+    /* A-law to u-law tests */
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        TESTDATA_DIR "DISK1/HOMING/16/I_INI_16.M",
+        TESTDATA_DIR "DISK2/HOMING/16/HN16FA.I",
+        TESTDATA_DIR "DISK2/HOMING/16/HN16FX.O",
+        16000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        TESTDATA_DIR "DISK1/HOMING/16/I_INI_16.M",
+        TESTDATA_DIR "DISK2/HOMING/16/HV16FA.I",
+        TESTDATA_DIR "DISK2/HOMING/16/HV16FX.O",
+        16000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        TESTDATA_DIR "DISK1/HOMING/24/I_INI_24.M",
+        TESTDATA_DIR "DISK2/HOMING/24/HN24FA.I",
+        TESTDATA_DIR "DISK2/HOMING/24/HN24FX.O",
+        24000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        TESTDATA_DIR "DISK1/HOMING/24/I_INI_24.M",
+        TESTDATA_DIR "DISK2/HOMING/24/HV24FA.I",
+        TESTDATA_DIR "DISK2/HOMING/24/HV24FX.O",
+        24000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        TESTDATA_DIR "DISK1/HOMING/32/I_INI_32.M",
+        TESTDATA_DIR "DISK2/HOMING/32/HN32FA.I",
+        TESTDATA_DIR "DISK2/HOMING/32/HN32FX.O",
+        32000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        TESTDATA_DIR "DISK1/HOMING/32/I_INI_32.M",
+        TESTDATA_DIR "DISK2/HOMING/32/HV32FA.I",
+        TESTDATA_DIR "DISK2/HOMING/32/HV32FX.O",
+        32000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/NRM.A",
+        TESTDATA_DIR "DISK1/HOMING/40/I_INI_40.M",
+        TESTDATA_DIR "DISK2/HOMING/40/HN40FA.I",
+        TESTDATA_DIR "DISK2/HOMING/40/HN40FX.O",
+        40000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        TESTDATA_DIR "DISK2/PCM_INIT.A",
+        TESTDATA_DIR "DISK2/INPUT/OVR.A",
+        TESTDATA_DIR "DISK1/HOMING/40/I_INI_40.M",
+        TESTDATA_DIR "DISK2/HOMING/40/HV40FA.I",
+        TESTDATA_DIR "DISK2/HOMING/40/HV40FX.O",
+        40000,
+        G726_ENCODING_ALAW,
+        G726_ENCODING_ULAW
+    },
+    {
+        NULL,
+        NULL,
+        NULL,
+        NULL,
+        NULL,
+        0,
+        0,
+        0
+    }
+};
+
+static int hex_get(char *s)
+{
+    int i;
+    int value;
+    int x;
+
+    for (value = i = 0;  i < 2;  i++)
+    {
+        x = *s++ - 0x30;
+        if (x > 9)
+            x -= 0x07;
+        if (x > 15)
+            x -= 0x20;
+        if (x < 0  ||  x > 15)
+            return -1;
+        value <<= 4;
+        value |= x;
+    }
+    return value;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int get_vector(FILE *file, uint8_t vec[])
+{
+    char buf[132 + 1];
+    char *s;
+    int i;
+    int value;
+
+    while (fgets(buf, 133, file))
+    {
+        s = buf;
+        i = 0;
+        while ((value = hex_get(s)) >= 0)
+        {
+            vec[i++] = value;
+            s += 2;
+        }
+        return i;
+    }
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int get_test_vector(const char *file, uint8_t buf[], int max_len)
+{
+    int octets;
+    int i;
+    int sum;
+    FILE *infile;
+    
+    if ((infile = fopen(file, "r")) == NULL)
+    {
+        fprintf(stderr, "    Failed to open '%s'\n", file);
+        exit(2);
+    }
+    octets = 0;  
+    while ((i = get_vector(infile, buf + octets)) > 0)
+        octets += i;
+    fclose(infile);
+    /* The last octet is a sumcheck, so the real data octets are one less than
+       the total we have */
+    octets--;
+    /* Test the checksum */
+    for (sum = i = 0;  i < octets;  i++)
+        sum += buf[i];
+    if (sum%255 != (int) buf[i])
+    {
+        fprintf(stderr, "    Sumcheck failed in '%s' - %x %x\n", file, sum%255, buf[i]);
+        exit(2);
+    }
+    return octets;
+}
+/*- End of function --------------------------------------------------------*/
+
+int main(int argc, char *argv[])
+{
+    g726_state_t enc_state;
+    g726_state_t dec_state;
+    int len2;
+    int len3;
+    int i;
+    int test;
+    int bits_per_code;
+    int itutests;
+    int bit_rate;
+    int bad_samples;
+    AFfilehandle inhandle;
+    AFfilehandle outhandle;
+    AFfilesetup filesetup;
+    int16_t amp[1024];
+    int frames;
+    int outframes;
+    int conditioning_samples;
+    int samples;
+    int conditioning_adpcm;
+    int adpcm;
+    int packing;
+    float x;
+
+    i = 1;
+    bit_rate = 32000;
+    itutests = TRUE;
+    packing = G726_PACKING_NONE;
+    while (argc > i)
+    {
+        if (strcmp(argv[i], "-16") == 0)
+        {
+            bit_rate = 16000;
+            itutests = FALSE;
+            i++;
+        }
+        else if (strcmp(argv[i], "-24") == 0)
+        {
+            bit_rate = 24000;
+            itutests = FALSE;
+            i++;
+        }
+        else if (strcmp(argv[i], "-32") == 0)
+        {
+            bit_rate = 32000;
+            itutests = FALSE;
+            i++;
+        }
+        else if (strcmp(argv[i], "-40") == 0)
+        {
+            bit_rate = 40000;
+            itutests = FALSE;
+            i++;
+        }
+        else if (strcmp(argv[i], "-l") == 0)
+        {
+            packing = G726_PACKING_LEFT;
+            i++;
+        }
+        else if (strcmp(argv[i], "-r") == 0)
+        {
+            packing = G726_PACKING_RIGHT;
+            i++;
+        }
+        else
+        {
+            fprintf(stderr, "Unknown parameter %s specified.\n", argv[i]);
+            exit(2);
+        }
+    }
+
+    len2 = 0;
+    conditioning_samples = 0;
+    if (itutests)
+    {
+        for (test = 0;  itu_test_sets[test].rate;  test++)
+        {
+            printf("Test %2d: '%s' + '%s'\n"
+                   "      -> '%s' + '%s'\n"
+                   "      -> '%s' [%d, %d, %d]\n",
+                   test,
+                   itu_test_sets[test].conditioning_pcm_file,
+                   itu_test_sets[test].pcm_file,
+                   itu_test_sets[test].conditioning_adpcm_file,
+                   itu_test_sets[test].adpcm_file,
+                   itu_test_sets[test].output_file,
+                   itu_test_sets[test].rate,
+                   itu_test_sets[test].compression_law,
+                   itu_test_sets[test].decompression_law);
+            switch (itu_test_sets[test].rate)
+            {
+            case 16000:
+                bits_per_code = 2;
+                break;
+            case 24000:
+                bits_per_code = 3;
+                break;
+            case 32000:
+            default:
+                bits_per_code = 4;
+                break;
+            case 40000:
+                bits_per_code = 5;
+                break;
+            }
+            if (itu_test_sets[test].compression_law != G726_ENCODING_NONE)
+            {
+                /* Test the encode side */
+                g726_init(&enc_state, itu_test_sets[test].rate, itu_test_sets[test].compression_law, G726_PACKING_NONE);
+                if (itu_test_sets[test].conditioning_pcm_file[0])
+                {
+                    conditioning_samples = get_test_vector(itu_test_sets[test].conditioning_pcm_file, xlaw, MAX_TEST_VECTOR_LEN);
+                    printf("Test %d: Homing %d samples at %dbps\n", test, conditioning_samples, itu_test_sets[test].rate);
+                }
+                else
+                {
+                    conditioning_samples = 0;
+                }
+                samples = get_test_vector(itu_test_sets[test].pcm_file, xlaw + conditioning_samples, MAX_TEST_VECTOR_LEN);
+                memcpy(itudata, xlaw, samples + conditioning_samples);
+                printf("Test %d: Compressing %d samples at %dbps\n", test, samples, itu_test_sets[test].rate);
+                len2 = g726_encode(&enc_state, adpcmdata, itudata, conditioning_samples + samples);
+            }
+            /* Test the decode side */
+            g726_init(&dec_state, itu_test_sets[test].rate, itu_test_sets[test].decompression_law, G726_PACKING_NONE);
+            if (itu_test_sets[test].conditioning_adpcm_file[0])
+            {
+                conditioning_adpcm = get_test_vector(itu_test_sets[test].conditioning_adpcm_file, unpacked, MAX_TEST_VECTOR_LEN);
+                printf("Test %d: Homing %d octets at %dbps\n", test, conditioning_adpcm, itu_test_sets[test].rate);
+            }
+            else
+            {
+                conditioning_adpcm = 0;
+            }
+            adpcm = get_test_vector(itu_test_sets[test].adpcm_file, unpacked + conditioning_adpcm, MAX_TEST_VECTOR_LEN);
+            if (itu_test_sets[test].compression_law != G726_ENCODING_NONE)
+            {
+                /* Test our compressed version against the reference compressed version */
+                printf("Test %d: Compressed data check - %d/%d octets\n", test, conditioning_adpcm + adpcm, len2);
+                if (conditioning_adpcm + adpcm == len2)
+                {
+                    for (bad_samples = 0, i = conditioning_samples;  i < len2;  i++)
+                    {
+                        if (adpcmdata[i] != unpacked[i])
+                        {
+                            bad_samples++;
+                            printf("Test %d: Compressed mismatch %d %x %x\n", test, i, adpcmdata[i], unpacked[i]);
+                        }
+                    }
+                    if (bad_samples > 0)
+                    {
+                        printf("Test failed\n");
+                        exit(2);
+                    }
+                    printf("Test passed\n");
+                }
+                else
+                {
+                    printf("Test %d: Length mismatch - ref = %d, processed = %d\n", test, conditioning_adpcm + adpcm, len2);
+                    exit(2);
+                }
+            }
+            
+            len3 = g726_decode(&dec_state, outdata, unpacked, conditioning_adpcm + adpcm);
+
+            /* Get the output reference data */
+            samples = get_test_vector(itu_test_sets[test].output_file, xlaw, MAX_TEST_VECTOR_LEN);
+            memcpy(itu_ref, xlaw, samples);
+            /* Test our decompressed version against the reference decompressed version */
+            printf("Test %d: Decompressed data check - %d/%d samples\n", test, samples, len3 - conditioning_adpcm);
+            if (samples == len3 - conditioning_adpcm)
+            {
+                for (bad_samples = 0, i = 0;  i < len3;  i++)
+                {
+                    if (itu_ref[i] != ((uint8_t *) outdata)[i + conditioning_adpcm])
+                    {
+                        bad_samples++;
+                        printf("Test %d: Decompressed mismatch %d %x %x\n", test, i, itu_ref[i], ((uint8_t *) outdata)[i + conditioning_adpcm]);
+                    }
+                }
+                if (bad_samples > 0)
+                {
+                    printf("Test failed\n");
+                    exit(2);
+                }
+                printf("Test passed\n");
+            }
+            else
+            {
+                printf("Test %d: Length mismatch - ref = %d, processed = %d\n", test, samples, len3 - conditioning_adpcm);
+                exit(2);
+            }
+        }
+
+        printf("Tests passed.\n");
+    }
+    else
+    {
+        if ((inhandle = afOpenFile(IN_FILE_NAME, "r", 0)) == AF_NULL_FILEHANDLE)
+        {
+            printf("    Cannot open wave file '%s'\n", IN_FILE_NAME);
+            exit(2);
+        }
+        if ((x = afGetFrameSize(inhandle, AF_DEFAULT_TRACK, 1)) != 2.0)
+        {
+            printf("    Unexpected frame size in wave file '%s'\n", IN_FILE_NAME);
+            exit(2);
+        }
+        if ((x = afGetRate(inhandle, AF_DEFAULT_TRACK)) != (float) SAMPLE_RATE)
+        {
+            printf("    Unexpected sample rate in wave file '%s'\n", IN_FILE_NAME);
+            exit(2);
+        }
+        if ((x = afGetChannels(inhandle, AF_DEFAULT_TRACK)) != 1.0)
+        {
+            printf("    Unexpected number of channels in wave file '%s'\n", IN_FILE_NAME);
+            exit(2);
+        }
+        if ((filesetup = afNewFileSetup()) == AF_NULL_FILESETUP)
+        {
+            fprintf(stderr, "    Failed to create file setup\n");
+            exit(2);
+        }
+        afInitSampleFormat(filesetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+        afInitRate(filesetup, AF_DEFAULT_TRACK, (float) SAMPLE_RATE);
+        afInitFileFormat(filesetup, AF_FILE_WAVE);
+        afInitChannels(filesetup, AF_DEFAULT_TRACK, 1);
+
+        outhandle = afOpenFile(OUT_FILE_NAME, "w", filesetup);
+        if (outhandle == AF_NULL_FILEHANDLE)
+        {
+            fprintf(stderr, "    Cannot create wave file '%s'\n", OUT_FILE_NAME);
+            exit(2);
+        }
+
+        printf("ADPCM packing is %d\n", packing);
+        g726_init(&enc_state, bit_rate, G726_ENCODING_LINEAR, packing);
+        g726_init(&dec_state, bit_rate, G726_ENCODING_LINEAR, packing);
+            
+        while ((frames = afReadFrames(inhandle, AF_DEFAULT_TRACK, amp, 159)))
+        {
+            adpcm = g726_encode(&enc_state, adpcmdata, amp, frames);
+            frames = g726_decode(&dec_state, amp, adpcmdata, adpcm);
+            outframes = afWriteFrames(outhandle, AF_DEFAULT_TRACK, amp, frames);
+        }
+        if (afCloseFile(inhandle) != 0)
+        {
+            printf("    Cannot close wave file '%s'\n", IN_FILE_NAME);
+            exit(2);
+        }
+        if (afCloseFile(outhandle) != 0)
+        {
+            printf("    Cannot close wave file '%s'\n", OUT_FILE_NAME);
+            exit(2);
+        }
+        afFreeFileSetup(filesetup);
+        printf("'%s' transcoded to '%s' at %dbps.\n", IN_FILE_NAME, OUT_FILE_NAME, bit_rate);
+    }
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/tests/gsm0610_tests.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/tests/gsm0610_tests.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,649 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * gsm0610_tests.c - Test the GSM 06.10 FR codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: gsm0610_tests.c,v 1.8 2007/11/10 11:14:58 steveu Exp $
+ */
+
+/*! \file */
+
+/*! \page gsm0610_tests_page GSM 06.10 full rate codec tests
+\section gsm0610_tests_page_sec_1 What does it do?
+Two sets of tests are performed:
+    - The tests defined in the GSM 06.10 specification, using the test data files supplied with
+      the specification.
+    - A generally audio quality test, consisting of compressing and decompressing a speeech
+      file for audible comparison.
+
+The speech file should be recorded at 16 bits/sample, 8000 samples/second, and named
+"pre_gsm0610.wav".
+
+\section gsm0610_tests_page_sec_2 How is it used?
+To perform the tests in the GSM 06.10 specification you need to obtain the test data files from the
+specification. These are copyright material, and so cannot be distributed with this test software.
+They can, however, be freely downloaded from the ETSI web site.
+
+The files, containing test vectors, which are supplied with the GSM 06.10 specification, should be
+copied to etsitests/gsm0610/unpacked so the files are arranged in the following directories.
+
+./fr_A:
+    Seq01-A.cod Seq01-A.inp Seq01-A.out
+    Seq02-A.cod Seq02-A.inp Seq02-A.out
+    Seq03-A.cod Seq03-A.inp Seq03-A.out
+    Seq04-A.cod Seq04-A.inp Seq04-A.out
+    Seq05-A.out
+
+./fr_L:
+    Seq01.cod   Seq01.inp   Seq01.out
+    Seq02.cod   Seq02.inp   Seq02.out
+    Seq03.cod   Seq03.inp   Seq03.out
+    Seq04.cod   Seq04.inp   Seq04.out
+    Seq05.cod   Seq05.out
+
+./fr_U:
+    Seq01-U.cod Seq01-U.inp Seq01-U.out
+    Seq02-U.cod Seq02-U.inp Seq02-U.out
+    Seq03-U.cod Seq03-U.inp Seq03-U.out
+    Seq04-U.cod Seq04-U.inp Seq04-U.out
+    Seq05-U.out
+
+./fr_homing_A:
+    Homing01_A.out
+    Seq01H_A.cod    Seq01H_A.inp    Seq01H_A.out
+    Seq02H_A.cod    Seq02H_A.inp    Seq02H_A.out
+    Seq03H_A.cod    Seq03H_A.inp    Seq03H_A.out
+    Seq04H_A.cod    Seq04H_A.inp    Seq04H_A.out
+    Seq05H_A.out
+    Seq06H_A.cod    Seq06H_A.inp
+
+./fr_homing_L:
+    Homing01.cod    Homing01.out
+    Seq01h.cod      Seq01h.inp      Seq01h.out
+    Seq02h.cod      Seq02h.inp      Seq02h.out
+    Seq03h.cod      Seq03h.inp      Seq03h.out
+    Seq04h.cod      Seq04h.inp      Seq04h.out
+    Seq05h.cod      Seq05h.out
+    Seq06h.cod      Seq06h.inp
+
+./fr_homing_U:
+    Homing01_U.out
+    Seq01H_U.cod    Seq01H_U.inp    Seq01H_U.out
+    Seq02H_U.cod    Seq02H_U.inp    Seq02H_U.out
+    Seq03H_U.cod    Seq03H_U.inp    Seq03H_U.out
+    Seq04H_U.cod    Seq04H_U.inp    Seq04H_U.out
+    Seq05H_U.out
+    Seq06H_U.cod    Seq06H_U.inp
+
+./fr_sync_A:
+    Seqsync_A.inp
+    Sync000_A.cod   --to--          Sync159_A.cod
+    
+./fr_sync_L:
+    Bitsync.inp
+    Seqsync.inp
+    Sync000.cod     --to--          Sync159.cod
+
+./fr_sync_U:
+    Seqsync_U.inp
+    Sync000_U.cod   --to--          Sync159_U.cod
+
+This is different from the directory structure in which they are supplied. Also, the files names are a little
+different. The supplied names are messy, and inconsistent across the sets. The names required by these tests
+just clean up these inconsistencies. Note that you will need a Windows machine to unpack some of the supplied
+files.
+
+To perform a general audio quality test, gsm0610_tests should be run. The file ../localtests/short_nb_voice.wav
+will be compressed to GSM 06.10 data, decompressed, and the resulting audio stored in post_gsm0610.wav.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <string.h>
+#include <ctype.h>
+#include <audiofile.h>
+
+#include "voipcodecs.h"
+
+#define BLOCK_LEN       160
+
+#define TEST_DIR "../etsitests/gsm0610/unpacked/fr_"
+
+#define IN_FILE_NAME    "../localtests/short_nb_voice.wav"
+#define OUT_FILE_NAME   "post_gsm0610.wav"
+
+#define HIST_LEN        1000
+
+uint8_t law_in_vector[1000000];
+int16_t in_vector[1000000];
+uint16_t code_vector_buf[1000000];
+uint8_t code_vector[1000000];
+uint8_t ref_code_vector[1000000];
+uint8_t decoder_code_vector[1000000];
+uint8_t law_out_vector[1000000];
+int16_t out_vector[1000000];
+int16_t ref_out_vector[1000000];
+uint8_t ref_law_out_vector[1000000];
+int vector_len;
+
+static int get_test_vector(int full, int disk, const char *name)
+{
+    char buf[500];
+    int in;
+    int len;
+    int i;
+    
+    if (full)
+    {
+        sprintf(buf, "%s%c/%s.inp", TEST_DIR, 'L', name);
+        if ((in = open(buf, O_RDONLY)) < 0)
+        {
+            fprintf(stderr, "Cannot open %s\n", buf);
+            exit(2);
+        }
+        len = read(in, in_vector, 1000000);
+        close(in);
+        len /= sizeof(int16_t);
+        vector_len = len;
+    }
+
+    sprintf(buf, "%s%c/%s.out", TEST_DIR, 'L', name);
+    if ((in = open(buf, O_RDONLY)) < 0)
+    {
+        fprintf(stderr, "Cannot open %s\n", buf);
+        exit(2);
+    }
+    len = read(in, ref_out_vector, 1000000);
+    close(in);
+    len /= sizeof(int16_t);
+    if (full)
+    {
+        if (len != vector_len)
+        {
+            fprintf(stderr, "Input and reference vector lengths do not match - %d %d\n", vector_len, len);
+            exit(2);
+        }
+    }
+    else
+    {
+        vector_len = len;
+    }
+    
+    sprintf(buf, "%s%c/%s.cod", TEST_DIR, 'L', name);
+    if ((in = open(buf, O_RDONLY)) < 0)
+    {
+        fprintf(stderr, "Cannot open %s\n", buf);
+        exit(2);
+    }
+    len = read(in, code_vector_buf, 1000000);
+    close(in);
+    len /= sizeof(int16_t);
+    for (i = 0;  i < len;  i++)
+    {
+        ref_code_vector[i] = code_vector_buf[i];
+        decoder_code_vector[i] = code_vector_buf[i];
+    }
+    if (len*BLOCK_LEN != vector_len*76)
+    {
+        fprintf(stderr, "Input and code vector lengths do not match - %d %d\n", vector_len, len);
+        exit(2);
+    }
+
+    return len;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int get_law_test_vector(int full, int law, const char *name)
+{
+    char buf[500];
+    int in;
+    int len;
+    int i;
+    int law_uc;
+    
+    law_uc = toupper(law);
+    
+    if (full)
+    {
+        sprintf(buf, "%s%c/%s-%c.inp", TEST_DIR, law_uc, name, law_uc);
+        if ((in = open(buf, O_RDONLY)) < 0)
+        {
+            fprintf(stderr, "Cannot open %s\n", buf);
+            exit(2);
+        }
+        len = read(in, law_in_vector, 1000000);
+        close(in);
+        vector_len = len;
+
+        sprintf(buf, "%s%c/%s-%c.cod", TEST_DIR, law_uc, name, law_uc);
+        if ((in = open(buf, O_RDONLY)) < 0)
+        {
+            fprintf(stderr, "Cannot open %s\n", buf);
+            exit(2);
+        }
+        len = read(in, code_vector_buf, 1000000);
+        close(in);
+        len /= sizeof(int16_t);
+        for (i = 0;  i < len;  i++)
+            ref_code_vector[i] = code_vector_buf[i];
+        if (len*BLOCK_LEN != vector_len*76)
+        {
+            fprintf(stderr, "Input and code vector lengths do not match - %d %d\n", vector_len, len);
+            exit(2);
+        }
+    }
+
+    sprintf(buf, "%s%c/%s-%c.out", TEST_DIR, law_uc, name, law_uc);
+    if ((in = open(buf, O_RDONLY)) < 0)
+    {
+        fprintf(stderr, "Cannot open %s\n", buf);
+        exit(2);
+    }
+    len = read(in, ref_law_out_vector, 1000000);
+    close(in);
+    if (full)
+    {
+        if (len != vector_len)
+        {
+            fprintf(stderr, "Input and reference vector lengths do not match - %d %d\n", vector_len, len);
+            exit(2);
+        }
+    }
+    else
+    {
+        vector_len = len;
+    }
+
+    sprintf(buf, "%s%c/%s.cod", TEST_DIR, 'L', name);
+    if ((in = open(buf, O_RDONLY)) < 0)
+    {
+        fprintf(stderr, "Cannot open %s\n", buf);
+        exit(2);
+    }
+    len = read(in, code_vector_buf, 1000000);
+    close(in);
+    len /= sizeof(int16_t);
+    for (i = 0;  i < len;  i++)
+        decoder_code_vector[i] = code_vector_buf[i];
+    
+    return len;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int perform_linear_test(int full, int disk, const char *name)
+{
+    gsm0610_state_t *gsm0610_enc_state;
+    gsm0610_state_t *gsm0610_dec_state;
+    int i;
+    int xxx;
+    int mismatches;
+
+    printf("Performing linear test '%s' from disk %d\n", name, disk);
+
+    get_test_vector(full, disk, name);
+
+    if (full)
+    {
+        if ((gsm0610_enc_state = gsm0610_init(NULL, GSM0610_PACKING_NONE)) == NULL)
+        {
+            fprintf(stderr, "    Cannot create encoder\n");
+            exit(2);
+        }
+        xxx = gsm0610_encode(gsm0610_enc_state, code_vector, in_vector, vector_len/BLOCK_LEN);
+
+        printf("Check code vector of length %d\n", xxx);
+        for (i = 0, mismatches = 0;  i < xxx;  i++)
+        {
+            if (code_vector[i] != ref_code_vector[i])
+            {
+                printf("%8d/%3d: %6d %6d\n", i/76, i%76, code_vector[i], ref_code_vector[i]);
+                mismatches++;
+            }
+        }
+        gsm0610_release(gsm0610_enc_state);
+        if (mismatches)
+        {
+            printf("Test failed: %d of %d samples mismatch\n", mismatches, xxx);
+            exit(2);
+        }
+        printf("Test passed\n");
+    }
+
+    if ((gsm0610_dec_state = gsm0610_init(NULL, GSM0610_PACKING_NONE)) == NULL)
+    {
+        fprintf(stderr, "    Cannot create decoder\n");
+        exit(2);
+    }
+    xxx = gsm0610_decode(gsm0610_dec_state, out_vector, decoder_code_vector, vector_len/BLOCK_LEN);
+    printf("Check output vector of length %d\n", vector_len);
+    for (i = 0, mismatches = 0;  i < vector_len;  i++)
+    {
+        if (out_vector[i] != ref_out_vector[i])
+        {
+            printf("%8d: %6d %6d\n", i, out_vector[i], ref_out_vector[i]);
+            mismatches++;
+        }
+    }
+    if (mismatches)
+    {
+        printf("Test failed: %d of %d samples mismatch\n", mismatches, vector_len);
+        exit(2);
+    }
+    gsm0610_release(gsm0610_dec_state);
+    printf("Test passed\n");
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int perform_law_test(int full, int law, const char *name)
+{
+    gsm0610_state_t *gsm0610_enc_state;
+    gsm0610_state_t *gsm0610_dec_state;
+    int i;
+    int xxx;
+    int mismatches;
+
+    if (law == 'a')
+        printf("Performing A-law test '%s'\n", name);
+    else
+        printf("Performing u-law test '%s'\n", name);
+        
+    get_law_test_vector(full, law, name);
+
+    if (full)
+    {
+        if ((gsm0610_enc_state = gsm0610_init(NULL, GSM0610_PACKING_NONE)) == NULL)
+        {
+            fprintf(stderr, "    Cannot create encoder\n");
+            exit(2);
+        }
+        if (law == 'a')
+        {
+            for (i = 0;  i < vector_len;  i++)
+                in_vector[i] = alaw_to_linear(law_in_vector[i]);
+        }
+        else
+        {
+            for (i = 0;  i < vector_len;  i++)
+                in_vector[i] = ulaw_to_linear(law_in_vector[i]);
+        }
+        xxx = gsm0610_encode(gsm0610_enc_state, code_vector, in_vector, vector_len/BLOCK_LEN);
+
+        printf("Check code vector of length %d\n", xxx);
+        for (i = 0, mismatches = 0;  i < xxx;  i++)
+        {
+            if (code_vector[i] != ref_code_vector[i])
+            {
+                printf("%8d/%3d: %6d %6d %6d\n", i/76, i%76, code_vector[i], ref_code_vector[i], decoder_code_vector[i]);
+                mismatches++;
+            }
+        }
+        if (mismatches)
+        {
+            printf("Test failed: %d of %d samples mismatch\n", mismatches, xxx);
+            exit(2);
+        }
+        printf("Test passed\n");
+        gsm0610_release(gsm0610_enc_state);
+    }
+
+    if ((gsm0610_dec_state = gsm0610_init(NULL, GSM0610_PACKING_NONE)) == NULL)
+    {
+        fprintf(stderr, "    Cannot create decoder\n");
+        exit(2);
+    }
+    xxx = gsm0610_decode(gsm0610_dec_state, out_vector, decoder_code_vector, vector_len/BLOCK_LEN);
+    if (law == 'a')
+    {
+        for (i = 0;  i < vector_len;  i++)
+            law_out_vector[i] = linear_to_alaw(out_vector[i]);
+    }
+    else
+    {
+        for (i = 0;  i < vector_len;  i++)
+            law_out_vector[i] = linear_to_ulaw(out_vector[i]);
+    }
+    printf("Check output vector of length %d\n", vector_len);
+    for (i = 0, mismatches = 0;  i < vector_len;  i++)
+    {
+        if (law_out_vector[i] != ref_law_out_vector[i])
+        {
+            printf("%8d: %6d %6d\n", i, law_out_vector[i], ref_law_out_vector[i]);
+            mismatches++;
+        }
+    }
+    if (mismatches)
+    {
+        printf("Test failed: %d of %d samples mismatch\n", mismatches, vector_len);
+        exit(2);
+    }
+    gsm0610_release(gsm0610_dec_state);
+    printf("Test passed\n");
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int repack_gsm0610_voip_to_wav49(uint8_t c[], const uint8_t d[])
+{
+    gsm0610_frame_t frame[2];
+    int n;
+ 
+	n = gsm0610_unpack_voip(&frame[0], d);
+	gsm0610_unpack_voip(&frame[1], d + n);
+    n = gsm0610_pack_wav49(c, frame);
+    return n;
+}
+/*- End of function --------------------------------------------------------*/
+
+static int repack_gsm0610_wav49_to_voip(uint8_t d[], const uint8_t c[])
+{
+    gsm0610_frame_t frame[2];
+    int n[2];
+
+    gsm0610_unpack_wav49(frame, c);
+    n[0] = gsm0610_pack_voip(d, &frame[0]);
+    n[1] = gsm0610_pack_voip(d + n[0], &frame[1]);
+    return n[0] + n[1];
+}
+/*- End of function --------------------------------------------------------*/
+
+static int perform_pack_unpack_test(void)
+{
+    uint8_t a[66];
+    uint8_t b[66];
+    uint8_t c[66];
+    int i;
+    int j;
+
+    printf("Performing packing/unpacking tests (not part of the ETSI conformance tests).\n");
+    /* Try trans-packing a lot of random data looking for before/after mismatch. */
+    for (j = 0;  j < 1000;  j++)
+    {        
+        for (i = 0;  i < 65;  i++)
+            a[i] = rand();
+        repack_gsm0610_wav49_to_voip(b, a);
+        repack_gsm0610_voip_to_wav49(c, b);
+        if (memcmp(a, c, 65))
+        {
+            printf("Test failed: data mismatch\n");
+            exit(2);
+        }
+
+        for (i = 0;  i < 66;  i++)
+            a[i] = rand();
+        /* Insert the magic code */
+        a[0] = (a[0] & 0xF) | 0xD0;
+        a[33] = (a[33] & 0xF) | 0xD0;
+        repack_gsm0610_voip_to_wav49(b, a);
+        repack_gsm0610_wav49_to_voip(c, b);
+        //for (i = 0;  i < 66;  i++)
+        //    printf("%2d: 0x%02X 0x%02X\n", i, a[i], c[i]);
+        if (memcmp(a, c, 66))
+        {
+            printf("Test failed: data mismatch\n");
+            exit(2);
+        }
+    }
+    printf("Test passed\n");
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+
+int main(int argc, char *argv[])
+{
+    AFfilehandle inhandle;
+    AFfilehandle outhandle;
+    AFfilesetup filesetup;
+    int frames;
+    int outframes;
+    float x;
+    int16_t pre_amp[HIST_LEN];
+    int16_t post_amp[HIST_LEN];
+    uint8_t gsm0610_data[HIST_LEN];
+    gsm0610_state_t *gsm0610_enc_state;
+    gsm0610_state_t *gsm0610_dec_state;
+    int etsitests;
+    int packing;
+    int i;
+
+    etsitests = TRUE;
+    packing = GSM0610_PACKING_NONE;
+    for (i = 1;  i < argc;  i++)
+    {
+        if (strcmp(argv[i], "-l") == 0)
+        {
+            etsitests = FALSE;
+            continue;
+        }
+        if (strcmp(argv[i], "-p") == 0)
+        {
+            packing = atoi(argv[++i]);
+            continue;
+        }
+        fprintf(stderr, "Unknown parameter %s specified.\n", argv[i]);
+        exit(2);
+    }
+
+    if (etsitests)
+    {
+        perform_linear_test(TRUE, 1, "Seq01");
+        perform_linear_test(TRUE, 1, "Seq02");
+        perform_linear_test(TRUE, 1, "Seq03");
+        perform_linear_test(TRUE, 1, "Seq04");
+        perform_linear_test(FALSE, 1, "Seq05");
+        perform_law_test(TRUE, 'a', "Seq01");
+        perform_law_test(TRUE, 'a', "Seq02");
+        perform_law_test(TRUE, 'a', "Seq03");
+        perform_law_test(TRUE, 'a', "Seq04");
+        perform_law_test(FALSE, 'a', "Seq05");
+        perform_law_test(TRUE, 'u', "Seq01");
+        perform_law_test(TRUE, 'u', "Seq02");
+        perform_law_test(TRUE, 'u', "Seq03");
+        perform_law_test(TRUE, 'u', "Seq04");
+        perform_law_test(FALSE, 'u', "Seq05");
+        /* This is not actually an ETSI test */
+        perform_pack_unpack_test();
+
+        printf("Tests passed.\n");
+    }
+    else
+    {
+        if ((inhandle = afOpenFile(IN_FILE_NAME, "r", 0)) == AF_NULL_FILEHANDLE)
+        {
+            printf("    Cannot open wave file '%s'\n", IN_FILE_NAME);
+            exit(2);
+        }
+        if ((x = afGetFrameSize(inhandle, AF_DEFAULT_TRACK, 1)) != 2.0)
+        {
+            printf("    Unexpected frame size in wave file '%s'\n", IN_FILE_NAME);
+            exit(2);
+        }
+        if ((x = afGetRate(inhandle, AF_DEFAULT_TRACK)) != (float) SAMPLE_RATE)
+        {
+            printf("    Unexpected sample rate in wave file '%s'\n", IN_FILE_NAME);
+            exit(2);
+        }
+        if ((x = afGetChannels(inhandle, AF_DEFAULT_TRACK)) != 1.0)
+        {
+            printf("    Unexpected number of channels in wave file '%s'\n", IN_FILE_NAME);
+            exit(2);
+        }
+        if ((filesetup = afNewFileSetup()) == AF_NULL_FILESETUP)
+        {
+            fprintf(stderr, "    Failed to create file setup\n");
+            exit(2);
+        }
+        afInitSampleFormat(filesetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+        afInitRate(filesetup, AF_DEFAULT_TRACK, (float) SAMPLE_RATE);
+        afInitFileFormat(filesetup, AF_FILE_WAVE);
+        afInitChannels(filesetup, AF_DEFAULT_TRACK, 1);
+    
+        outhandle = afOpenFile(OUT_FILE_NAME, "w", filesetup);
+        if (outhandle == AF_NULL_FILEHANDLE)
+        {
+            fprintf(stderr, "    Cannot create wave file '%s'\n", OUT_FILE_NAME);
+            exit(2);
+        }
+    
+        if ((gsm0610_enc_state = gsm0610_init(NULL, packing)) == NULL)
+        {
+            fprintf(stderr, "    Cannot create encoder\n");
+            exit(2);
+        }
+            
+        if ((gsm0610_dec_state = gsm0610_init(NULL, packing)) == NULL)
+        {
+            fprintf(stderr, "    Cannot create decoder\n");
+            exit(2);
+        }
+
+        while ((frames = afReadFrames(inhandle, AF_DEFAULT_TRACK, pre_amp, 2*BLOCK_LEN)))
+        {
+            gsm0610_encode(gsm0610_enc_state, gsm0610_data, pre_amp, (packing == GSM0610_PACKING_WAV49)  ?  1  :  2);
+            gsm0610_decode(gsm0610_dec_state, post_amp, gsm0610_data, (packing == GSM0610_PACKING_WAV49)  ?  1  :  2);
+            outframes = afWriteFrames(outhandle, AF_DEFAULT_TRACK, post_amp, frames);
+        }
+    
+        if (afCloseFile(inhandle) != 0)
+        {
+            printf("    Cannot close wave file '%s'\n", IN_FILE_NAME);
+            exit(2);
+        }
+        if (afCloseFile(outhandle) != 0)
+        {
+            printf("    Cannot close wave file '%s'\n", OUT_FILE_NAME);
+            exit(2);
+        }
+        afFreeFileSetup(filesetup);
+        gsm0610_release(gsm0610_enc_state);
+        gsm0610_release(gsm0610_dec_state);
+    }
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/tests/ima_adpcm_tests.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/tests/ima_adpcm_tests.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,206 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * ima_adpcm_tests.c - Test the IMA/DVI/Intel ADPCM encode and decode
+ *                     software.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2004 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id$
+ */
+
+/*! \file */
+
+/*! \page ima_adpcm_tests_page IMA ADPCM tests
+\section ima_adpcm_tests_page_sec_1 What does it do?
+To perform a general audio quality test, ima_adpcm_tests should be run. The test file
+../localtests/short_nb_voice.wav will be compressed to the specified bit rate,
+decompressed, and the resulting audio stored in post_ima_adpcm.wav. A simple SNR test
+is automatically performed. Listening tests may be used for a more detailed evaluation
+of the degradation in quality caused by the compression.
+
+\section ima_adpcm_tests_page_sec_2 How is it used?
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <fcntl.h>
+#include <string.h>
+#include <time.h>
+#include <audiofile.h>
+
+#include "voipcodecs.h"
+
+#define IN_FILE_NAME    "../localtests/short_nb_voice.wav"
+#define OUT_FILE_NAME   "post_ima_adpcm.wav"
+
+#define HIST_LEN        1000
+
+int main(int argc, char *argv[])
+{
+    int i;
+    AFfilehandle inhandle;
+    AFfilehandle outhandle;
+    AFfilesetup filesetup;
+    int frames;
+    int dec_frames;
+    int outframes;
+    int ima_bytes;
+    float x;
+    double pre_energy;
+    double post_energy;
+    double diff_energy;
+    int16_t pre_amp[HIST_LEN];
+    int16_t post_amp[HIST_LEN];
+    uint8_t ima_data[HIST_LEN];
+    int16_t history[HIST_LEN];
+    int hist_in;
+    int hist_out;
+    ima_adpcm_state_t *ima_enc_state;
+    ima_adpcm_state_t *ima_dec_state;
+    int xx;
+    int vbr;
+    const char *in_file_name;
+
+    vbr = FALSE;
+    in_file_name = IN_FILE_NAME;
+    for (i = 1;  i < argc;  i++)
+    {
+        if (strcmp(argv[i], "-v") == 0)
+        {
+            vbr = TRUE;
+            continue;
+        }
+        if (strcmp(argv[i], "-i") == 0)
+        {
+            in_file_name = argv[++i];
+            continue;
+        }
+        fprintf(stderr, "Unknown parameter %s specified.\n", argv[i]);
+        exit(2);
+    }
+
+    if ((inhandle = afOpenFile(in_file_name, "r", 0)) == AF_NULL_FILEHANDLE)
+    {
+        printf("    Cannot open wave file '%s'\n", in_file_name);
+        exit(2);
+    }
+    if ((x = afGetFrameSize(inhandle, AF_DEFAULT_TRACK, 1)) != 2.0)
+    {
+        printf("    Unexpected frame size in wave file '%s'\n", in_file_name);
+        exit(2);
+    }
+    if ((x = afGetRate(inhandle, AF_DEFAULT_TRACK)) != (float) SAMPLE_RATE)
+    {
+        printf("    Unexpected sample rate in wave file '%s'\n", in_file_name);
+        exit(2);
+    }
+    if ((x = afGetChannels(inhandle, AF_DEFAULT_TRACK)) != 1.0)
+    {
+        printf("    Unexpected number of channels in wave file '%s'\n", in_file_name);
+        exit(2);
+    }
+
+    if ((filesetup = afNewFileSetup()) == AF_NULL_FILESETUP)
+    {
+        fprintf(stderr, "    Failed to create file setup\n");
+        exit(2);
+    }
+    afInitSampleFormat(filesetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+    afInitRate(filesetup, AF_DEFAULT_TRACK, (float) SAMPLE_RATE);
+    afInitFileFormat(filesetup, AF_FILE_WAVE);
+    afInitChannels(filesetup, AF_DEFAULT_TRACK, 1);
+    if ((outhandle = afOpenFile(OUT_FILE_NAME, "w", filesetup)) == AF_NULL_FILEHANDLE)
+    {
+        fprintf(stderr, "    Cannot create wave file '%s'\n", OUT_FILE_NAME);
+        exit(2);
+    }
+
+    if ((ima_enc_state = ima_adpcm_init(NULL, (vbr)  ?  IMA_ADPCM_VDVI  :  IMA_ADPCM_DVI4)) == NULL)
+    {
+        fprintf(stderr, "    Cannot create encoder\n");
+        exit(2);
+    }
+        
+    if ((ima_dec_state = ima_adpcm_init(NULL, (vbr)  ?  IMA_ADPCM_VDVI  :  IMA_ADPCM_DVI4)) == NULL)
+    {
+        fprintf(stderr, "    Cannot create decoder\n");
+        exit(2);
+    }
+
+    hist_in = 0;
+    hist_out = 0;
+    pre_energy = 0.0;
+    post_energy = 0.0;
+    diff_energy = 0.0;
+    while ((frames = afReadFrames(inhandle, AF_DEFAULT_TRACK, pre_amp, 159)))
+    {
+        ima_bytes = ima_adpcm_encode(ima_enc_state, ima_data, pre_amp, frames);
+        dec_frames = ima_adpcm_decode(ima_dec_state, post_amp, ima_data, ima_bytes);
+        for (i = 0;  i < frames;  i++)
+        {
+            history[hist_in++] = pre_amp[i];
+            if (hist_in >= HIST_LEN)
+                hist_in = 0;
+            pre_energy += (double) pre_amp[i] * (double) pre_amp[i];
+        }
+        for (i = 0;  i < dec_frames;  i++)
+        {
+            post_energy += (double) post_amp[i] * (double) post_amp[i];
+            xx = post_amp[i] - history[hist_out++];
+            if (hist_out >= HIST_LEN)
+                hist_out = 0;
+            diff_energy += (double) xx * (double) xx;
+        }
+        outframes = afWriteFrames(outhandle, AF_DEFAULT_TRACK, post_amp, dec_frames);
+    }
+    if (afCloseFile(inhandle) != 0)
+    {
+        printf("    Cannot close wave file '%s'\n", in_file_name);
+        exit(2);
+    }
+    if (afCloseFile(outhandle) != 0)
+    {
+        printf("    Cannot close wave file '%s'\n", OUT_FILE_NAME);
+        exit(2);
+    }
+    afFreeFileSetup(filesetup);
+    ima_adpcm_release(ima_enc_state);
+    ima_adpcm_release(ima_dec_state);
+
+    printf("Output energy is %f%% of input energy.\n", 100.0*post_energy/pre_energy);
+    printf("Residual energy is %f%% of the total.\n", 100.0*diff_energy/post_energy);
+    if (fabs(1.0 - post_energy/pre_energy) > 0.05
+        ||
+        fabs(diff_energy/post_energy) > 0.03)
+    {
+        printf("Tests failed.\n");
+        exit(2);
+    }
+    
+    printf("Tests passed.\n");
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/tests/lpc10_tests.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/tests/lpc10_tests.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,307 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * lpc10_tests.c - Test the LPC10 low bit rate speech codec.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2006 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: lpc10_tests.c,v 1.11 2007/11/10 11:14:58 steveu Exp $
+ */
+
+/*! \file */
+
+/*! \page lpc10_tests_page LPC10 codec tests
+\section lpc10_tests_page_sec_1 What does it do?
+
+\section lpc10_tests_page_sec_2 How is it used?
+To perform a general audio quality test, lpc10 should be run. The file ../localtests/short_nb_voice.wav
+will be compressed to LPC10 data, decompressed, and the resulting audio stored in post_lpc10.wav.
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <fcntl.h>
+#include <unistd.h>
+#include <string.h>
+#include <assert.h>
+#include <ctype.h>
+#include <audiofile.h>
+
+#include "voipcodecs.h"
+
+#define BLOCK_LEN       180
+
+#define BLOCKS_PER_READ 1
+
+#define IN_FILE_NAME            "../localtests/dam9.wav"
+#define REF_FILE_NAME           "../localtests/dam9_lpc55.wav"
+#define COMPRESS_FILE_NAME      "lpc10_out.lpc10"
+#define DECOMPRESS_FILE_NAME    "lpc10_in.lpc10"
+#define OUT_FILE_NAME           "post_lpc10.wav"
+
+int main(int argc, char *argv[])
+{
+    AFfilehandle inhandle;
+    AFfilehandle refhandle;
+    AFfilehandle outhandle;
+    AFfilesetup filesetup;
+    int frames;
+    int outframes;
+    float x;
+    double pre_energy;
+    double post_energy;
+    double ref_energy;
+    double diff_energy;
+    int16_t pre_amp[BLOCKS_PER_READ*BLOCK_LEN];
+    int16_t post_amp[BLOCKS_PER_READ*BLOCK_LEN];
+    int16_t ref_amp[BLOCKS_PER_READ*BLOCK_LEN];
+    int16_t log_amp[BLOCKS_PER_READ*BLOCK_LEN*3];
+    uint8_t lpc10_data[BLOCKS_PER_READ*7];
+    double xx;
+    lpc10_encode_state_t *lpc10_enc_state;
+    lpc10_decode_state_t *lpc10_dec_state;
+    int i;
+    int block_no;
+    int log_error;
+    int compress;
+    int decompress;
+    const char *in_file_name;
+    int compress_file;
+    int decompress_file;
+    int len;
+
+    compress = FALSE;
+    decompress = FALSE;
+    log_error = TRUE;
+    in_file_name = IN_FILE_NAME;
+    for (i = 1;  i < argc;  i++)
+    {
+        if (strcmp(argv[i], "-c") == 0)
+        {
+            compress = TRUE;
+            continue;
+        }
+        if (strcmp(argv[i], "-d") == 0)
+        {
+            decompress = TRUE;
+            continue;
+        }
+        if (strcmp(argv[i], "-i") == 0)
+        {
+            in_file_name = argv[++i];
+            continue;
+        }
+        if (strcmp(argv[i], "-l") == 0)
+        {
+            log_error = FALSE;
+            continue;
+        }
+    }
+
+    compress_file = -1;
+    decompress_file = -1;
+    inhandle = AF_NULL_FILEHANDLE;
+    refhandle = AF_NULL_FILEHANDLE;
+    outhandle = AF_NULL_FILEHANDLE;
+    if (!decompress)
+    {
+        if ((inhandle = afOpenFile(in_file_name, "r", 0)) == AF_NULL_FILEHANDLE)
+        {
+            printf("    Cannot open wave file '%s'\n", in_file_name);
+            exit(2);
+        }
+        if ((x = afGetFrameSize(inhandle, AF_DEFAULT_TRACK, 1)) != 2.0)
+        {
+            printf("    Unexpected frame size in wave file '%s'\n", in_file_name);
+            exit(2);
+        }
+        if ((x = afGetRate(inhandle, AF_DEFAULT_TRACK)) != (float) SAMPLE_RATE)
+        {
+            printf("    Unexpected sample rate in wave file '%s'\n", in_file_name);
+            exit(2);
+        }
+        if ((x = afGetChannels(inhandle, AF_DEFAULT_TRACK)) != 1.0)
+        {
+            printf("    Unexpected number of channels in wave file '%s'\n", in_file_name);
+            exit(2);
+        }
+        if ((filesetup = afNewFileSetup()) == AF_NULL_FILESETUP)
+        {
+            fprintf(stderr, "    Failed to create file setup\n");
+            exit(2);
+        }
+
+        if ((refhandle = afOpenFile(REF_FILE_NAME, "r", 0)) == AF_NULL_FILEHANDLE)
+        {
+            printf("    Cannot open wave file '%s'\n", REF_FILE_NAME);
+            exit(2);
+        }
+        if ((x = afGetFrameSize(refhandle, AF_DEFAULT_TRACK, 1)) != 2.0)
+        {
+            printf("    Unexpected frame size in wave file '%s'\n", REF_FILE_NAME);
+            exit(2);
+        }
+        if ((x = afGetRate(refhandle, AF_DEFAULT_TRACK)) != (float) SAMPLE_RATE)
+        {
+            printf("    Unexpected sample rate in wave file '%s'\n", REF_FILE_NAME);
+            exit(2);
+        }
+        if ((x = afGetChannels(refhandle, AF_DEFAULT_TRACK)) != 1.0)
+        {
+            printf("    Unexpected number of channels in wave file '%s'\n", REF_FILE_NAME);
+            exit(2);
+        }
+    }
+    else
+    {
+        if ((decompress_file = open(DECOMPRESS_FILE_NAME, O_RDONLY)) < 0)
+        {
+            fprintf(stderr, "    Cannot open decompressed data file '%s'\n", DECOMPRESS_FILE_NAME);
+            exit(2);
+        }
+    }
+
+    if ((filesetup = afNewFileSetup()) == AF_NULL_FILESETUP)
+    {
+        fprintf(stderr, "    Failed to create file setup\n");
+        exit(2);
+    }
+    afInitSampleFormat(filesetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+    afInitRate(filesetup, AF_DEFAULT_TRACK, (float) SAMPLE_RATE);
+    afInitFileFormat(filesetup, AF_FILE_WAVE);
+    afInitChannels(filesetup, AF_DEFAULT_TRACK, 1);
+
+    if ((outhandle = afOpenFile(OUT_FILE_NAME, "w", filesetup)) == AF_NULL_FILEHANDLE)
+    {
+        fprintf(stderr, "    Cannot create wave file '%s'\n", OUT_FILE_NAME);
+        exit(2);
+    }
+    
+    if ((lpc10_enc_state = lpc10_encode_init(NULL, TRUE)) == NULL)
+    {
+        fprintf(stderr, "    Cannot create encoder\n");
+        exit(2);
+    }
+            
+    if ((lpc10_dec_state = lpc10_decode_init(NULL, TRUE)) == NULL)
+    {
+        fprintf(stderr, "    Cannot create decoder\n");
+        exit(2);
+    }
+
+    if (compress)
+    {
+        if ((compress_file = open(COMPRESS_FILE_NAME, O_WRONLY | O_CREAT | O_TRUNC, 0666)) < 0)
+        {
+            fprintf(stderr, "    Cannot create compressed data file '%s'\n", COMPRESS_FILE_NAME);
+            exit(2);
+        }
+    }
+
+    pre_energy = 0.0;
+    post_energy = 0.0;
+    ref_energy = 0.0;
+    diff_energy = 0.0;
+
+    if (decompress)
+    {
+        while ((len = read(decompress_file, lpc10_data, BLOCKS_PER_READ*7)) > 0)
+        {
+            lpc10_decode(lpc10_dec_state, post_amp, lpc10_data, len/7);
+            outframes = afWriteFrames(outhandle, AF_DEFAULT_TRACK, post_amp, BLOCK_LEN*len/7);
+        }
+    }
+    else
+    {
+        block_no = 0;
+        while ((frames = afReadFrames(inhandle, AF_DEFAULT_TRACK, pre_amp, BLOCKS_PER_READ*BLOCK_LEN)) == BLOCKS_PER_READ*BLOCK_LEN
+                &&
+                (frames = afReadFrames(refhandle, AF_DEFAULT_TRACK, ref_amp, BLOCKS_PER_READ*BLOCK_LEN)) == BLOCKS_PER_READ*BLOCK_LEN)
+        {
+            lpc10_encode(lpc10_enc_state, lpc10_data, pre_amp, BLOCKS_PER_READ);
+            if (compress)
+                write(compress_file, lpc10_data, BLOCKS_PER_READ*7);
+            lpc10_decode(lpc10_dec_state, post_amp, lpc10_data, BLOCKS_PER_READ);
+            for (i = 0;  i < BLOCK_LEN;  i++)
+            {
+                pre_energy += (double) pre_amp[i]*(double) pre_amp[i];
+                post_energy += (double) post_amp[i]*(double) post_amp[i];
+                ref_energy += (double) ref_amp[i]*(double) ref_amp[i];
+                /* The reference file has some odd clipping, so eliminate this from the
+                   energy measurement. */
+                if (ref_amp[i] == 32767  ||  ref_amp[i] == -32768)
+                    xx = 0.0;
+                else
+                    xx = post_amp[i] - ref_amp[i];
+                diff_energy += (double) xx*(double) xx;
+                log_amp[i] = xx;
+            }
+            block_no++;
+            if (log_error)
+                outframes = afWriteFrames(outhandle, AF_DEFAULT_TRACK, log_amp, frames);
+            else
+                outframes = afWriteFrames(outhandle, AF_DEFAULT_TRACK, post_amp, frames);
+        }
+        if (afCloseFile(inhandle) != 0)
+        {
+            printf("    Cannot close wave file '%s'\n", in_file_name);
+            exit(2);
+        }
+        if (afCloseFile(refhandle) != 0)
+        {
+            printf("    Cannot close wave file '%s'\n", REF_FILE_NAME);
+            exit(2);
+        }
+    }
+    
+    if (afCloseFile(outhandle) != 0)
+    {
+        printf("    Cannot close wave file '%s'\n", OUT_FILE_NAME);
+        exit(2);
+    }
+    afFreeFileSetup(filesetup);
+    if (compress)
+        close(compress_file);
+    if (decompress)
+        close(decompress_file);
+    lpc10_encode_release(lpc10_enc_state);
+    lpc10_decode_release(lpc10_dec_state);
+
+    if (!decompress)
+    {
+        printf("Output energy is %f%% of input energy.\n", 100.0*post_energy/pre_energy);
+        printf("Difference energy is %f%% of the total.\n", 100.0*diff_energy/ref_energy);
+        if (fabs(1.0 - post_energy/pre_energy) > 0.05
+            ||
+            fabs(diff_energy/post_energy) > 0.03)
+        {
+            printf("Tests failed.\n");
+            exit(2);
+        }
+        printf("Tests passed.\n");
+    }
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/tests/oki_adpcm_tests.c
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/tests/oki_adpcm_tests.c	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,239 @@
+/*
+ * VoIPcodecs - a series of DSP components for telephony
+ *
+ * oki_adpcm_tests.c - Test the Oki (Dialogic) ADPCM encode and decode
+ *                     software at 24kbps and 32kbps.
+ *
+ * Written by Steve Underwood <steveu@coppice.org>
+ *
+ * Copyright (C) 2004 Steve Underwood
+ *
+ * All rights reserved.
+ *
+ * This program is free software; you can redistribute it and/or modify
+ * it under the terms of the GNU General Public License version 2, as
+ * published by the Free Software Foundation.
+ *
+ * This program is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+ * GNU General Public License for more details.
+ *
+ * You should have received a copy of the GNU General Public License
+ * along with this program; if not, write to the Free Software
+ * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+ *
+ * $Id: oki_adpcm_tests.c,v 1.24 2007/11/10 11:14:58 steveu Exp $
+ */
+
+/*! \file */
+
+/*! \page oki_adpcm_tests_page OKI (Dialogic) ADPCM tests
+\section oki_adpcm_tests_page_sec_1 What does it do?
+To perform a general audio quality test, oki_adpcm_tests should be run. The test file
+../localtests/short_nb_voice.wav will be compressed to the specified bit rate,
+decompressed, and the resulting audio stored in post_oki_adpcm.wav. A simple SNR test
+is automatically performed. Listening tests may be used for a more detailed evaluation
+of the degradation in quality caused by the compression. Both 32k bps and 24k bps
+compression may be tested.
+
+\section oki_adpcm_tests_page_sec_2 How is it used?
+*/
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <fcntl.h>
+#include <string.h>
+#include <time.h>
+#include <audiofile.h>
+
+#include "voipcodecs.h"
+
+#define IN_FILE_NAME    "../localtests/short_nb_voice.wav"
+#define OUT_FILE_NAME   "post_oki_adpcm.wav"
+
+#define HIST_LEN        1000
+
+int main(int argc, char *argv[])
+{
+    int i;
+    AFfilehandle inhandle;
+    AFfilehandle outhandle;
+    AFfilesetup filesetup;
+    int frames;
+    int dec_frames;
+    int outframes;
+    int oki_bytes;
+    int bit_rate;
+    float x;
+    double pre_energy;
+    double post_energy;
+    double diff_energy;
+    int16_t pre_amp[HIST_LEN];
+    int16_t post_amp[HIST_LEN];
+    uint8_t oki_data[HIST_LEN];
+    int16_t history[HIST_LEN];
+    int hist_in;
+    int hist_out;
+    oki_adpcm_state_t *oki_enc_state;
+    oki_adpcm_state_t *oki_dec_state;
+    int xx;
+    int total_pre_samples;
+    int total_compressed_bytes;
+    int total_post_samples;
+    const char *in_file_name;
+
+    bit_rate = 32000;
+    in_file_name = IN_FILE_NAME;
+    for (i = 1;  i < argc;  i++)
+    {
+        if (strcmp(argv[i], "-2") == 0)
+        {
+            bit_rate = 24000;
+            continue;
+        }
+        if (strcmp(argv[i], "-i") == 0)
+        {
+            in_file_name = argv[++i];
+            continue;
+        }
+        fprintf(stderr, "Unknown parameter %s specified.\n", argv[i]);
+        exit(2);
+    }
+
+    if ((inhandle = afOpenFile(in_file_name, "r", 0)) == AF_NULL_FILEHANDLE)
+    {
+        printf("    Cannot open wave file '%s'\n", in_file_name);
+        exit(2);
+    }
+    if ((x = afGetFrameSize(inhandle, AF_DEFAULT_TRACK, 1)) != 2.0)
+    {
+        printf("    Unexpected frame size in wave file '%s'\n", in_file_name);
+        exit(2);
+    }
+    if ((x = afGetRate(inhandle, AF_DEFAULT_TRACK)) != (float) SAMPLE_RATE)
+    {
+        printf("    Unexpected sample rate in wave file '%s'\n", in_file_name);
+        exit(2);
+    }
+    if ((x = afGetChannels(inhandle, AF_DEFAULT_TRACK)) != 1.0)
+    {
+        printf("    Unexpected number of channels in wave file '%s'\n", in_file_name);
+        exit(2);
+    }
+    if ((filesetup = afNewFileSetup()) == AF_NULL_FILESETUP)
+    {
+        fprintf(stderr, "    Failed to create file setup\n");
+        exit(2);
+    }
+    afInitSampleFormat(filesetup, AF_DEFAULT_TRACK, AF_SAMPFMT_TWOSCOMP, 16);
+    afInitRate(filesetup, AF_DEFAULT_TRACK, (float) SAMPLE_RATE);
+    afInitFileFormat(filesetup, AF_FILE_WAVE);
+    afInitChannels(filesetup, AF_DEFAULT_TRACK, 1);
+
+    outhandle = afOpenFile(OUT_FILE_NAME, "w", filesetup);
+    if (outhandle == AF_NULL_FILEHANDLE)
+    {
+        fprintf(stderr, "    Cannot create wave file '%s'\n", OUT_FILE_NAME);
+        exit(2);
+    }
+
+    if ((oki_enc_state = oki_adpcm_init(NULL, bit_rate)) == NULL)
+    {
+        fprintf(stderr, "    Cannot create encoder\n");
+        exit(2);
+    }
+        
+    if ((oki_dec_state = oki_adpcm_init(NULL, bit_rate)) == NULL)
+    {
+        fprintf(stderr, "    Cannot create decoder\n");
+        exit(2);
+    }
+
+    hist_in = 0;
+    if (bit_rate == 32000)
+        hist_out = 0;
+    else
+        hist_out = HIST_LEN - 27;
+    memset(history, 0, sizeof(history));
+    pre_energy = 0.0;
+    post_energy = 0.0;
+    diff_energy = 0.0;
+    total_pre_samples = 0;
+    total_compressed_bytes = 0;
+    total_post_samples = 0;
+    while ((frames = afReadFrames(inhandle, AF_DEFAULT_TRACK, pre_amp, 159)))
+    {
+        total_pre_samples +=frames;
+        oki_bytes = oki_adpcm_encode(oki_enc_state, oki_data, pre_amp, frames);
+        total_compressed_bytes += oki_bytes;
+        dec_frames = oki_adpcm_decode(oki_dec_state, post_amp, oki_data, oki_bytes);
+        total_post_samples += dec_frames;
+        for (i = 0;  i < frames;  i++)
+        {
+            history[hist_in++] = pre_amp[i];
+            if (hist_in >= HIST_LEN)
+                hist_in = 0;
+            pre_energy += (double) pre_amp[i] * (double) pre_amp[i];
+        }
+        for (i = 0;  i < dec_frames;  i++)
+        {
+            post_energy += (double) post_amp[i] * (double) post_amp[i];
+            xx = post_amp[i] - history[hist_out++];
+            if (hist_out >= HIST_LEN)
+                hist_out = 0;
+            diff_energy += (double) xx * (double) xx;
+            //post_amp[i] = xx;
+        }
+        outframes = afWriteFrames(outhandle, AF_DEFAULT_TRACK, post_amp, dec_frames);
+    }
+    if (afCloseFile(inhandle) != 0)
+    {
+        printf("    Cannot close wave file '%s'\n", in_file_name);
+        exit(2);
+    }
+    if (afCloseFile(outhandle) != 0)
+    {
+        printf("    Cannot close wave file '%s'\n", OUT_FILE_NAME);
+        exit(2);
+    }
+    afFreeFileSetup(filesetup);
+    oki_adpcm_release(oki_enc_state);
+    oki_adpcm_release(oki_dec_state);
+
+    printf("Pre samples: %d\n", total_pre_samples);
+    printf("Compressed bytes: %d\n", total_compressed_bytes);
+    printf("Post samples: %d\n", total_post_samples);
+
+    printf("Output energy is %f%% of input energy.\n", 100.0*post_energy/pre_energy);
+    printf("Residual energy is %f%% of the total.\n", 100.0*diff_energy/post_energy);
+    if (bit_rate == 32000)
+    {
+        if (fabs(1.0 - post_energy/pre_energy) > 0.05
+            ||
+            fabs(diff_energy/post_energy) > 0.03)
+        {
+            printf("Tests failed.\n");
+            exit(2);
+        }
+    }
+    else
+    {
+        if (fabs(1.0 - post_energy/pre_energy) > 0.20
+            ||
+            fabs(diff_energy/post_energy) > 0.10)
+        {
+            printf("Tests failed.\n");
+            exit(2);
+        }
+    }
+
+    printf("Tests passed.\n");
+    return 0;
+}
+/*- End of function --------------------------------------------------------*/
+/*- End of file ------------------------------------------------------------*/

Added: freeswitch/trunk/libs/voipcodecs/tests/regression_tests.sh
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/tests/regression_tests.sh	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,93 @@
+#!/bin/sh
+#
+# VoIPcodecs - a series of DSP components for telephony
+#
+# regression_tests.sh
+#
+# This program is free software; you can redistribute it and/or modify
+# it under the terms of the GNU General Public License version 2, as
+# published by the Free Software Foundation.
+#
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY; without even the implied warranty of
+# MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the
+# GNU General Public License for more details.
+#
+# You should have received a copy of the GNU General Public License
+# along with this program; if not, write to the Free Software
+# Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+#
+# $Id: regression_tests.sh,v 1.47 2007/12/22 12:37:22 steveu Exp $
+#
+
+STDOUT_DEST=xyzzy
+STDERR_DEST=xyzzy2
+
+echo Performing basic VoIP codecs regression tests
+echo
+
+./g711_tests >$STDOUT_DEST 2>$STDERR_DEST
+RETVAL=$?
+if [ $RETVAL != 0 ]
+then
+    echo g711_tests failed!
+    exit $RETVAL
+fi
+echo g711_tests completed OK
+
+./g722_tests >$STDOUT_DEST 2>$STDERR_DEST
+RETVAL=$?
+if [ $RETVAL != 0 ]
+then
+    echo g722_tests failed!
+    exit $RETVAL
+fi
+echo g722_tests completed OK
+
+./g726_tests >$STDOUT_DEST 2>$STDERR_DEST
+RETVAL=$?
+if [ $RETVAL != 0 ]
+then
+    echo g726_tests failed!
+    exit $RETVAL
+fi
+echo g726_tests completed OK
+
+./gsm0610_tests >$STDOUT_DEST 2>$STDERR_DEST
+RETVAL=$?
+if [ $RETVAL != 0 ]
+then
+    echo gsm0610_tests failed!
+    exit $RETVAL
+fi
+echo gsm0610_tests completed OK
+
+./ima_adpcm_tests >$STDOUT_DEST 2>$STDERR_DEST
+RETVAL=$?
+if [ $RETVAL != 0 ]
+then
+    echo ima_adpcm_tests failed!
+    exit $RETVAL
+fi
+echo ima_adpcm_tests completed OK
+
+./lpc10_tests >$STDOUT_DEST 2>$STDERR_DEST
+RETVAL=$?
+if [ $RETVAL != 0 ]
+then
+    echo lpc10_tests failed!
+    exit $RETVAL
+fi
+echo lpc10_tests completed OK
+
+./oki_adpcm_tests >$STDOUT_DEST 2>$STDERR_DEST
+RETVAL=$?
+if [ $RETVAL != 0 ]
+then
+    echo oki_adpcm_tests failed!
+    exit $RETVAL
+fi
+echo oki_adpcm_tests completed OK
+
+echo
+echo All regression tests successfully completed

Added: freeswitch/trunk/libs/voipcodecs/voipcodecs.spec.in
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/voipcodecs.spec.in	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,65 @@
+Summary:    VoIPcodecs is a library of unencumbered codecs commonly used for VoIP
+Name:       @PACKAGE@
+Version:    @VERSION@
+Release:    1
+License:    GPL
+Group:      System Environment/Libraries
+URL:        http://www.soft-switch.org/voipcodecs
+BuildRoot:  %{_tmppath}/%{name}-%{version}-root
+Source:     http://www.soft-switch.org/downloads/voipcodecs/@PACKAGE@-@VERSION@.tar.gz
+BuildRoot:  %{_tmppath}/%{name}-%{version}-%{release}-root-%(%{__id_u} -n)
+
+Docdir:     %{_prefix}/doc
+
+BuildRequires: audiofile-devel
+BuildRequires: doxygen
+
+%description
+VoIPcodecs is a library of unencumbered codecs commonly used for VoIP.
+
+%package devel
+Summary:    VoIPcodecs development files
+Group:      Development/Libraries
+Requires:   voipcodecs = %{version}
+PreReq:     /sbin/install-info
+
+%description devel
+VoIPcodecs development files.
+
+%prep
+%setup -q
+
+%build
+%configure --enable-doc --disable-static --disable-rpath
+make
+
+%install
+rm -rf %{buildroot}
+make install DESTDIR=%{buildroot}
+rm %{buildroot}%{_libdir}/libvoipcodecs.la
+
+%clean
+rm -rf %{buildroot}
+
+%files
+%defattr(-,root,root,-)
+%doc ChangeLog AUTHORS COPYING NEWS README 
+
+%{_libdir}/libvoipcodecs.so.*
+
+%{_datadir}/voipcodecs
+
+%files devel
+%defattr(-,root,root,-)
+%doc doc/api
+%{_includedir}/voipcodecs.h
+%{_includedir}/voipcodecs
+%{_libdir}/libvoipcodecs.so
+
+%post -p /sbin/ldconfig
+
+%postun -p /sbin/ldconfig
+
+%changelog
+* Thu Feb  7 2008 Steve Underwood <steveu@coppice.org> 0.0.1
+- First pass

Added: freeswitch/trunk/libs/voipcodecs/wrapper.xsl
==============================================================================
--- (empty file)
+++ freeswitch/trunk/libs/voipcodecs/wrapper.xsl	Thu Feb  7 17:00:10 2008
@@ -0,0 +1,5 @@
+<xsl:stylesheet xmlns:xsl="http://www.w3.org/1999/XSL/Transform"
+                version='1.0'>
+  <xsl:import href="http://docbook.sourceforge.net/release/xsl/current/xhtml/chunk.xsl"/>
+  <xsl:param name="html.stylesheet">css.css</xsl:param>
+</xsl:stylesheet>
\ No newline at end of file

    

[Freeswitch-svn] [commit] r7544 - in freeswitch/trunk/libs/voipcodecs: . config debian doc etsitests itutests localtests src src/msvc src/msvc/sys src/voipcodecs tests

Freeswitch SVN