Working codec2 support

bootstrap.sh

@@ -16,7 +16,7 @@ done
 BASEDIR=`pwd`;
 LIBDIR=${BASEDIR}/libs;
 SUBDIRS="ilbc curl iksemel js js/nsprpub libdingaling libedit libsndfile pcre sofia-sip \
-        speex sqlite srtp openzap freetdm spandsp libg722_1 portaudio unimrcp tiff-3.8.2 broadvoice silk";
+        speex sqlite srtp openzap freetdm spandsp libg722_1 portaudio unimrcp tiff-3.8.2 broadvoice silk libcodec2";
 
 if [ ! -f modules.conf ]; then 
     cp build/modules.conf.in modules.conf

configure.in

@@ -998,6 +998,7 @@ AC_CONFIG_SUBDIRS([libs/spandsp])
 AC_CONFIG_SUBDIRS([libs/broadvoice])
 AC_CONFIG_SUBDIRS([libs/libg722_1])
 AC_CONFIG_SUBDIRS([libs/silk])
+AC_CONFIG_SUBDIRS([libs/libcodec2])
 
 case $host in
    *-openbsd*)

libs/libcodec2/COPYING

@@ -0,0 +1,502 @@
+                  GNU LESSER GENERAL PUBLIC LICENSE
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libs/libcodec2/INSTALL

@@ -0,0 +1,236 @@
+Installation Instructions
+*************************
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+
+Compilers and Options
+=====================
+
+Some systems require unusual options for compilation or linking that the
+`configure' script does not know about.  Run `./configure --help' for
+details on some of the pertinent environment variables.
+
+   You can give `configure' initial values for configuration parameters
+by setting variables in the command line or in the environment.  Here
+is an example:
+
+     ./configure CC=c89 CFLAGS=-O2 LIBS=-lposix
+
+   *Note Defining Variables::, for more details.
+
+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 support 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' installs the package's commands under
+`/usr/local/bin', include files under `/usr/local/include', etc.  You
+can specify an installation prefix other than `/usr/local' by giving
+`configure' the option `--prefix=PREFIX'.
+
+   You can specify separate installation prefixes for
+architecture-specific files and architecture-independent files.  If you
+pass the option `--exec-prefix=PREFIX' to `configure', the package uses
+PREFIX as the prefix for installing programs and libraries.
+Documentation and other data files still use the regular prefix.
+
+   In addition, if you use an unusual directory layout you can give
+options like `--bindir=DIR' 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' cannot figure out automatically,
+but needs to determine by the type of machine the package will run on.
+Usually, assuming the package is built to be run on the _same_
+architectures, `configure' can figure that out, but if it prints a
+message saying it cannot guess the machine type, give it the
+`--build=TYPE' option.  TYPE can either be a short name for the system
+type, such as `sun4', or a canonical name which has the form:
+
+     CPU-COMPANY-SYSTEM
+
+where SYSTEM can have one of these forms:
+
+     OS KERNEL-OS
+
+   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 machine type.
+
+   If you are _building_ compiler tools for cross-compiling, you should
+use the option `--target=TYPE' to select the type of system they will
+produce code for.
+
+   If you want to _use_ a cross compiler, that generates code for a
+platform different from the build platform, you should specify the
+"host" platform (i.e., that on which the generated programs will
+eventually be run) with `--host=TYPE'.
+
+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.
+
+Defining Variables
+==================
+
+Variables not defined in a site shell script can be set in the
+environment passed to `configure'.  However, some packages may run
+configure again during the build, and the customized values of these
+variables may be lost.  In order to avoid this problem, you should set
+them in the `configure' command line, using `VAR=value'.  For example:
+
+     ./configure CC=/usr/local2/bin/gcc
+
+causes the specified `gcc' to be used as the C compiler (unless it is
+overridden in the site shell script).  Here is a another example:
+
+     /bin/bash ./configure CONFIG_SHELL=/bin/bash
+
+Here the `CONFIG_SHELL=/bin/bash' operand causes subsequent
+configuration-related scripts to be executed by `/bin/bash'.
+
+`configure' Invocation
+======================
+
+`configure' recognizes the following options to control how it operates.
+
+`--help'
+`-h'
+     Print a summary of the options to `configure', and exit.
+
+`--version'
+`-V'
+     Print the version of Autoconf used to generate the `configure'
+     script, and exit.
+
+`--cache-file=FILE'
+     Enable the cache: use and save the results of the tests in FILE,
+     traditionally `config.cache'.  FILE defaults to `/dev/null' to
+     disable caching.
+
+`--config-cache'
+`-C'
+     Alias for `--cache-file=config.cache'.
+
+`--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.
+
+`configure' also accepts some other, not widely useful, options.  Run
+`configure --help' for more details.
+

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+	rev="$$rev ."; \
+	target=`echo $@ | sed s/-recursive//`; \
+	for subdir in $$rev; do \
+	  echo "Making $$target in $$subdir"; \
+	  if test "$$subdir" = "."; then \
+	    local_target="$$target-am"; \
+	  else \
+	    local_target="$$target"; \
+	  fi; \
+	  (cd $$subdir && $(MAKE) $(AM_MAKEFLAGS) $$local_target) \
+	  || eval $$failcom; \
+	done && test -z "$$fail"
+tags-recursive:
+	list='$(SUBDIRS)'; for subdir in $$list; do \
+	  test "$$subdir" = . || (cd $$subdir && $(MAKE) $(AM_MAKEFLAGS) tags); \
+	done
+ctags-recursive:
+	list='$(SUBDIRS)'; for subdir in $$list; do \
+	  test "$$subdir" = . || (cd $$subdir && $(MAKE) $(AM_MAKEFLAGS) ctags); \
+	done
+
+ID: $(HEADERS) $(SOURCES) $(LISP) $(TAGS_FILES)
+	list='$(SOURCES) $(HEADERS) $(LISP) $(TAGS_FILES)'; \
+	unique=`for i in $$list; do \
+	    if test -f "$$i"; then echo $$i; else echo $(srcdir)/$$i; fi; \
+	  done | \
+	  $(AWK) '    { files[$$0] = 1; } \
+	       END { for (i in files) print i; }'`; \
+	mkid -fID $$unique
+tags: TAGS
+
+TAGS: tags-recursive $(HEADERS) $(SOURCES)  $(TAGS_DEPENDENCIES) \
+		$(TAGS_FILES) $(LISP)
+	tags=; \
+	here=`pwd`; \
+	if ($(ETAGS) --etags-include --version) >/dev/null 2>&1; then \
+	  include_option=--etags-include; \
+	  empty_fix=.; \
+	else \
+	  include_option=--include; \
+	  empty_fix=; \
+	fi; \
+	list='$(SUBDIRS)'; for subdir in $$list; do \
+	  if test "$$subdir" = .; then :; else \
+	    test ! -f $$subdir/TAGS || \
+	      tags="$$tags $$include_option=$$here/$$subdir/TAGS"; \
+	  fi; \
+	done; \
+	list='$(SOURCES) $(HEADERS)  $(LISP) $(TAGS_FILES)'; \
+	unique=`for i in $$list; do \
+	    if test -f "$$i"; then echo $$i; else echo $(srcdir)/$$i; fi; \
+	  done | \
+	  $(AWK) '    { files[$$0] = 1; } \
+	       END { for (i in files) print i; }'`; \
+	if test -z "$(ETAGS_ARGS)$$tags$$unique"; then :; else \
+	  test -n "$$unique" || unique=$$empty_fix; \
+	  $(ETAGS) $(ETAGSFLAGS) $(AM_ETAGSFLAGS) $(ETAGS_ARGS) \
+	    $$tags $$unique; \
+	fi
+ctags: CTAGS
+CTAGS: ctags-recursive $(HEADERS) $(SOURCES)  $(TAGS_DEPENDENCIES) \
+		$(TAGS_FILES) $(LISP)
+	tags=; \
+	here=`pwd`; \
+	list='$(SOURCES) $(HEADERS)  $(LISP) $(TAGS_FILES)'; \
+	unique=`for i in $$list; do \
+	    if test -f "$$i"; then echo $$i; else echo $(srcdir)/$$i; fi; \
+	  done | \
+	  $(AWK) '    { files[$$0] = 1; } \
+	       END { for (i in files) print i; }'`; \
+	test -z "$(CTAGS_ARGS)$$tags$$unique" \
+	  || $(CTAGS) $(CTAGSFLAGS) $(AM_CTAGSFLAGS) $(CTAGS_ARGS) \
+	     $$tags $$unique
+
+GTAGS:
+	here=`$(am__cd) $(top_builddir) && pwd` \
+	  && cd $(top_srcdir) \
+	  && gtags -i $(GTAGS_ARGS) $$here
+
+distclean-tags:
+	-rm -f TAGS ID GTAGS GRTAGS GSYMS GPATH tags
+
+distdir: $(DISTFILES)
+	$(am__remove_distdir)
+	mkdir $(distdir)
+	$(mkdir_p) $(distdir)/doc $(distdir)/octave $(distdir)/pitch $(distdir)/raw $(distdir)/script $(distdir)/src $(distdir)/unittest $(distdir)/wav
+	@srcdirstrip=`echo "$(srcdir)" | sed 's|.|.|g'`; \
+	topsrcdirstrip=`echo "$(top_srcdir)" | sed 's|.|.|g'`; \
+	list='$(DISTFILES)'; for file in $$list; do \
+	  case $$file in \
+	    $(srcdir)/*) file=`echo "$$file" | sed "s|^$$srcdirstrip/||"`;; \
+	    $(top_srcdir)/*) file=`echo "$$file" | sed "s|^$$topsrcdirstrip/|$(top_builddir)/|"`;; \
+	  esac; \
+	  if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+	  dir=`echo "$$file" | sed -e 's,/[^/]*$$,,'`; \
+	  if test "$$dir" != "$$file" && test "$$dir" != "."; then \
+	    dir="/$$dir"; \
+	    $(mkdir_p) "$(distdir)$$dir"; \
+	  else \
+	    dir=''; \
+	  fi; \
+	  if test -d $$d/$$file; then \
+	    if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+	      cp -pR $(srcdir)/$$file $(distdir)$$dir || exit 1; \
+	    fi; \
+	    cp -pR $$d/$$file $(distdir)$$dir || exit 1; \
+	  else \
+	    test -f $(distdir)/$$file \
+	    || cp -p $$d/$$file $(distdir)/$$file \
+	    || exit 1; \
+	  fi; \
+	done
+	list='$(DIST_SUBDIRS)'; for subdir in $$list; do \
+	  if test "$$subdir" = .; then :; else \
+	    test -d "$(distdir)/$$subdir" \
+	    || $(mkdir_p) "$(distdir)/$$subdir" \
+	    || exit 1; \
+	    distdir=`$(am__cd) $(distdir) && pwd`; \
+	    top_distdir=`$(am__cd) $(top_distdir) && pwd`; \
+	    (cd $$subdir && \
+	      $(MAKE) $(AM_MAKEFLAGS) \
+	        top_distdir="$$top_distdir" \
+	        distdir="$$distdir/$$subdir" \
+	        distdir) \
+	      || exit 1; \
+	  fi; \
+	done
+	-find $(distdir) -type d ! -perm -777 -exec chmod a+rwx {} \; -o \
+	  ! -type d ! -perm -444 -links 1 -exec chmod a+r {} \; -o \
+	  ! -type d ! -perm -400 -exec chmod a+r {} \; -o \
+	  ! -type d ! -perm -444 -exec $(SHELL) $(install_sh) -c -m a+r {} {} \; \
+	|| chmod -R a+r $(distdir)
+dist-gzip: distdir
+	tardir=$(distdir) && $(am__tar) | GZIP=$(GZIP_ENV) gzip -c >$(distdir).tar.gz
+	$(am__remove_distdir)
+
+dist-bzip2: distdir
+	tardir=$(distdir) && $(am__tar) | bzip2 -9 -c >$(distdir).tar.bz2
+	$(am__remove_distdir)
+
+dist-tarZ: distdir
+	tardir=$(distdir) && $(am__tar) | compress -c >$(distdir).tar.Z
+	$(am__remove_distdir)
+
+dist-shar: distdir
+	shar $(distdir) | GZIP=$(GZIP_ENV) gzip -c >$(distdir).shar.gz
+	$(am__remove_distdir)
+
+dist-zip: distdir
+	-rm -f $(distdir).zip
+	zip -rq $(distdir).zip $(distdir)
+	$(am__remove_distdir)
+
+dist dist-all: distdir
+	tardir=$(distdir) && $(am__tar) | GZIP=$(GZIP_ENV) gzip -c >$(distdir).tar.gz
+	$(am__remove_distdir)
+
+# This target untars the dist file and tries a VPATH configuration.  Then
+# it guarantees that the distribution is self-contained by making another
+# tarfile.
+distcheck: dist
+	case '$(DIST_ARCHIVES)' in \
+	*.tar.gz*) \
+	  GZIP=$(GZIP_ENV) gunzip -c $(distdir).tar.gz | $(am__untar) ;;\
+	*.tar.bz2*) \
+	  bunzip2 -c $(distdir).tar.bz2 | $(am__untar) ;;\
+	*.tar.Z*) \
+	  uncompress -c $(distdir).tar.Z | $(am__untar) ;;\
+	*.shar.gz*) \
+	  GZIP=$(GZIP_ENV) gunzip -c $(distdir).shar.gz | unshar ;;\
+	*.zip*) \
+	  unzip $(distdir).zip ;;\
+	esac
+	chmod -R a-w $(distdir); chmod a+w $(distdir)
+	mkdir $(distdir)/_build
+	mkdir $(distdir)/_inst
+	chmod a-w $(distdir)
+	dc_install_base=`$(am__cd) $(distdir)/_inst && pwd | sed -e 's,^[^:\\/]:[\\/],/,'` \
+	  && dc_destdir="$${TMPDIR-/tmp}/am-dc-$$$$/" \
+	  && cd $(distdir)/_build \
+	  && ../configure --srcdir=.. --prefix="$$dc_install_base" \
+	    $(DISTCHECK_CONFIGURE_FLAGS) \
+	  && $(MAKE) $(AM_MAKEFLAGS) \
+	  && $(MAKE) $(AM_MAKEFLAGS) dvi \
+	  && $(MAKE) $(AM_MAKEFLAGS) check \
+	  && $(MAKE) $(AM_MAKEFLAGS) install \
+	  && $(MAKE) $(AM_MAKEFLAGS) installcheck \
+	  && $(MAKE) $(AM_MAKEFLAGS) uninstall \
+	  && $(MAKE) $(AM_MAKEFLAGS) distuninstallcheck_dir="$$dc_install_base" \
+	        distuninstallcheck \
+	  && chmod -R a-w "$$dc_install_base" \
+	  && ({ \
+	       (cd ../.. && umask 077 && mkdir "$$dc_destdir") \
+	       && $(MAKE) $(AM_MAKEFLAGS) DESTDIR="$$dc_destdir" install \
+	       && $(MAKE) $(AM_MAKEFLAGS) DESTDIR="$$dc_destdir" uninstall \
+	       && $(MAKE) $(AM_MAKEFLAGS) DESTDIR="$$dc_destdir" \
+	            distuninstallcheck_dir="$$dc_destdir" distuninstallcheck; \
+	      } || { rm -rf "$$dc_destdir"; exit 1; }) \
+	  && rm -rf "$$dc_destdir" \
+	  && $(MAKE) $(AM_MAKEFLAGS) dist \
+	  && rm -rf $(DIST_ARCHIVES) \
+	  && $(MAKE) $(AM_MAKEFLAGS) distcleancheck
+	$(am__remove_distdir)
+	@(echo "$(distdir) archives ready for distribution: "; \
+	  list='$(DIST_ARCHIVES)'; for i in $$list; do echo $$i; done) | \
+	  sed -e '1{h;s/./=/g;p;x;}' -e '$${p;x;}'
+distuninstallcheck:
+	@cd $(distuninstallcheck_dir) \
+	&& test `$(distuninstallcheck_listfiles) | wc -l` -le 1 \
+	   || { echo "ERROR: files left after uninstall:" ; \
+	        if test -n "$(DESTDIR)"; then \
+	          echo "  (check DESTDIR support)"; \
+	        fi ; \
+	        $(distuninstallcheck_listfiles) ; \
+	        exit 1; } >&2
+distcleancheck: distclean
+	@if test '$(srcdir)' = . ; then \
+	  echo "ERROR: distcleancheck can only run from a VPATH build" ; \
+	  exit 1 ; \
+	fi
+	@test `$(distcleancheck_listfiles) | wc -l` -eq 0 \
+	  || { echo "ERROR: files left in build directory after distclean:" ; \
+	       $(distcleancheck_listfiles) ; \
+	       exit 1; } >&2
+check-am: all-am
+check: check-recursive
+all-am: Makefile
+installdirs: installdirs-recursive
+installdirs-am:
+install: install-recursive
+install-exec: install-exec-recursive
+install-data: install-data-recursive
+uninstall: uninstall-recursive
+
+install-am: all-am
+	@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
+
+installcheck: installcheck-recursive
+install-strip:
+	$(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	  install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	  `test -z '$(STRIP)' || \
+	    echo "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'"` install
+mostlyclean-generic:
+
+clean-generic:
+
+distclean-generic:
+	-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+
+maintainer-clean-generic:
+	@echo "This command is intended for maintainers to use"
+	@echo "it deletes files that may require special tools to rebuild."
+clean: clean-recursive
+
+clean-am: clean-generic clean-libtool mostlyclean-am
+
+distclean: distclean-recursive
+	-rm -f $(am__CONFIG_DISTCLEAN_FILES)
+	-rm -f Makefile
+distclean-am: clean-am distclean-generic distclean-libtool \
+	distclean-tags
+
+dvi: dvi-recursive
+
+dvi-am:
+
+html: html-recursive
+
+info: info-recursive
+
+info-am:
+
+install-data-am:
+
+install-exec-am:
+
+install-info: install-info-recursive
+
+install-man:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-recursive
+	-rm -f $(am__CONFIG_DISTCLEAN_FILES)
+	-rm -rf $(top_srcdir)/autom4te.cache
+	-rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-recursive
+
+mostlyclean-am: mostlyclean-generic mostlyclean-libtool
+
+pdf: pdf-recursive
+
+pdf-am:
+
+ps: ps-recursive
+
+ps-am:
+
+uninstall-am: uninstall-info-am
+
+uninstall-info: uninstall-info-recursive
+
+.PHONY: $(RECURSIVE_TARGETS) CTAGS GTAGS all all-am am--refresh check \
+	check-am clean clean-generic clean-libtool clean-recursive \
+	ctags ctags-recursive dist dist-all dist-bzip2 dist-gzip \
+	dist-shar dist-tarZ dist-zip distcheck distclean \
+	distclean-generic distclean-libtool distclean-recursive \
+	distclean-tags distcleancheck distdir distuninstallcheck dvi \
+	dvi-am html html-am info info-am install install-am \
+	install-data install-data-am install-exec install-exec-am \
+	install-info install-info-am install-man install-strip \
+	installcheck installcheck-am installdirs installdirs-am \
+	maintainer-clean maintainer-clean-generic \
+	maintainer-clean-recursive mostlyclean mostlyclean-generic \
+	mostlyclean-libtool mostlyclean-recursive pdf pdf-am ps ps-am \
+	tags tags-recursive uninstall uninstall-am uninstall-info-am
+
+# Tell versions [3.59,3.63) of GNU make to not export all variables.
+# Otherwise a system limit (for SysV at least) may be exceeded.
+.NOEXPORT:

libs/libcodec2/configure.gnu

@@ -0,0 +1,4 @@
+#! /bin/sh
+srcpath=$(dirname $0 2>/dev/null )  || srcpath="." 
+$srcpath/configure "$@" --disable-shared --with-pic
+

libs/libcodec2/configure.in

@@ -0,0 +1,26 @@
+#                                               -*- Autoconf -*-
+# Process this file with autoconf to produce a configure script.
+
+AC_PREREQ([2.59])
+AC_INIT(libcodec2, 1.0, david@rowetel.com)
+AM_INIT_AUTOMAKE(libcodec2,1.0)
+
+# Checks for programs.
+AC_PROG_CC
+AC_PROG_LIBTOOL
+
+# Checks for libraries.
+# FIXME: Replace `main' with a function in `-lm':
+AC_CHECK_LIB([m], [main])
+
+# Checks for header files.
+AC_CHECK_HEADERS([stdlib.h string.h])
+
+# Checks for typedefs, structures, and compiler characteristics.
+
+# Checks for library functions.
+AC_FUNC_MALLOC
+AC_CHECK_FUNCS([floor pow sqrt])
+
+AC_CONFIG_FILES([Makefile src/Makefile unittest/Makefile])
+AC_OUTPUT

libs/libcodec2/doc/s_n.txt

@@ -0,0 +1 @@
+s(n)=A_1cos(\omega_0+\phi_1)+A_2cos(2\omega_0+\phi_2)+...+A_Lcos(L\omega_0+\phi_L)

libs/libcodec2/octave/glottal.m

@@ -0,0 +1,25 @@
+% glottal.m
+% David Rowe 12 Sep 2009
+% Matlab script to generate the phase spectra of a glottal pulse
+
+% lpc10 pulse from spandsp.  When the file glottal.c was used as a part of the
+% excitation phase component in phase.c, phase_synth_zero_order(), no difference 
+% in speech quality was apparent.  So left out of code for now.
+
+sh=12
+kexc = [ 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];
+kexc = shift(kexc,sh);
+kexc = [kexc(1:sh) zeros(1,512-25) kexc(sh+1:25)];
+figure(1)
+plot(kexc)
+figure(2)
+G = fft(kexc);
+plot((1:256)*(4000/256),unwrap(angle(G(1:256))))
+
+f=fopen("glottal.c","wt");
+fprintf(f,"float glottal[]={\n");
+for m=1:255
+  fprintf(f,"  %f,\n",angle(G(m)));
+endfor
+fprintf(f,"  %f};\n",angle(G(256)));
+fclose(f);

libs/libcodec2/octave/load_raw.m

@@ -0,0 +1,8 @@
+% load_raw.m
+% David Rowe 7 Oct 2009
+
+function s = load_raw(fn)
+  fs=fopen(fn,"rb");
+  s = fread(fs,Inf,"short");
+  plot(s)
+endfunction

libs/libcodec2/octave/lsp_pdf.m

@@ -0,0 +1,50 @@
+% lsp_pdf.m
+% David Rowe 2 Oct 2009
+% Plots histograms (PDF estimates) of LSP training data
+
+function lsp_pdf(lsp)
+  [r,c] = size(lsp);
+
+  % LSPs
+
+  figure(3);
+  clf;
+  [x,y] = hist(lsp(:,1),100);
+  plot(y*4000/pi,x,";1;");
+  hold on;
+  for i=2:c
+    [x,y] = hist(lsp(:,i),100);
+    legend = sprintf(";%d;",i);
+    plot(y*4000/pi,x,legend);
+  endfor
+  hold off;
+  grid;
+
+  % LSP differences
+
+  figure(4);
+  clf;
+  subplot(211)
+  [x,y] = hist(lsp(:,1),100);
+  plot(y,x,";1;");
+  hold on;
+  for i=2:5
+    [x,y] = hist(lsp(:,i) - lsp(:,i-1),100);
+    legend = sprintf(";%d;",i);
+    plot(y,x,legend);
+  endfor
+  hold off;
+  grid;
+
+  subplot(212)
+  [x,y] = hist(lsp(:,6)-lsp(:,5),100);
+  plot(y,x,";6;");
+  hold on;
+  for i=7:c
+    [x,y] = hist(lsp(:,i) - lsp(:,i-1),100);
+    legend = sprintf(";%d;",i);
+    plot(y,x,legend);
+  endfor
+  hold off;
+  grid;
+endfunction

libs/libcodec2/octave/phase.m

@@ -0,0 +1,56 @@
+% phase.m
+% David Rowe August 2009
+% experiments with phase for sinusoidal codecs
+
+function phase(samname, F0, png)
+  Wo=2*pi*F0/8000;
+  P=2*pi/Wo;
+  L = floor(pi/Wo);
+  Nsam = 16000;
+  N = 80;
+  F = Nsam/N;
+  A = 10000/L;
+  phi = zeros(1,L);
+  s = zeros(1,Nsam);
+  
+  for m=floor(L/2):L
+    phi_off(m) = -m*Wo*8;
+  end
+
+  for f=1:F
+    phi(1) = phi(1) + Wo*N;
+    phi(1) = mod(phi(1),2*pi);
+  
+    for m=1:L
+      phi(m) = m*phi(1);
+    end
+
+    x = zeros(1,N);
+    for m=1:L
+      x = x + A*cos(m*Wo*(0:(N-1)) + phi(m));
+    endfor
+    s((f-1)*N+1:f*N) = x;
+  endfor
+
+  figure(1);
+  clf;
+  plot(s(1:250));
+
+  fs=fopen(samname,"wb");
+  fwrite(fs,s,"short");
+  fclose(fs);
+
+  if (nargin == 3)
+      % small image to fit blog
+
+      __gnuplot_set__ terminal png size 450,300
+      ss = sprintf("__gnuplot_set__ output \"%s.png\"", samname);
+      eval(ss)
+      replot;
+
+      % for some reason I need this to stop large plot getting wiped
+      __gnuplot_set__ output "/dev/null"
+  endif
+
+endfunction
+

libs/libcodec2/octave/phase2.m

@@ -0,0 +1,50 @@
+% phase2.m
+% David Rowe Sep 2009
+% experiments with phase for sinusoidal codecs, looking at phase
+% of excitation with real Am samples from hts1
+
+function phase2(samname, png)
+  N = 16000;
+
+  f=45;
+  model = load("../src/hts1a_model.txt");
+  phase = load("../src/hts1a_phase_phase.txt");
+  Wo = model(f,1);
+  P=2*pi/Wo;
+  L = model(f,2);
+  A = model(f,3:(L+2));
+  phi = phase(f,1:L);
+  phi = zeros(1,L);
+  for m=L/2:L
+    phi(m) = 2*pi*rand(1,1);
+  end
+
+  s = zeros(1,N);
+
+  for m=1:L
+    s_m = A(m)*cos(m*Wo*(0:(N-1)) + phi(m));
+    s = s + s_m;
+  endfor
+
+  figure(1);
+  clf;
+  plot(s(1:250));
+
+  fs=fopen(samname,"wb");
+  fwrite(fs,s,"short");
+  fclose(fs);
+
+  if (nargin == 2)
+      % small image to fit blog
+
+      __gnuplot_set__ terminal png size 450,300
+      ss = sprintf("__gnuplot_set__ output \"%s.png\"", samname);
+      eval(ss)
+      replot;
+
+      % for some reason I need this to stop large plot getting wiped
+      __gnuplot_set__ output "/dev/null"
+  endif
+
+endfunction
+

libs/libcodec2/octave/pitch_test.m

@@ -0,0 +1,39 @@
+% pitch_test.m
+% David Rowe Sep 2009
+% Constructs a sequence to test the pitch estimator
+
+function pitch_test(samname)
+  M=320;
+  F=200;
+
+  fs=fopen(samname,"wb");
+
+  f0 = 100;
+  for f=1:200
+    Wo=2*pi*f0/8000;
+    P=2*pi/Wo;
+    L = floor(pi/Wo);
+    A = 10000/L;
+    phi = zeros(1,L);
+    s = zeros(1,M);
+
+    for m=1:L
+      s = s + A*cos(m*Wo*(0:(M-1)) + phi(m));
+    endfor
+
+    figure(1);
+    clf;
+    plot(s);
+
+    fwrite(fs,s,"short");
+
+    f0 = f0 + 5;
+    if (f0 > 400)
+      f0 = 100;    
+    endif
+  endfor
+
+  fclose(fs);
+
+endfunction
+

libs/libcodec2/octave/pl.m

@@ -0,0 +1,42 @@
+% Copyright David Rowe 2009
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+
+function pl(samname1, start_sam, end_sam, pngname)
+  
+  fs=fopen(samname1,"rb");
+  s=fread(fs,Inf,"short");
+
+  st = 1;
+  en = length(s);
+  if (nargin >= 2)
+    st = start_sam;
+  endif
+  if (nargin >= 3)
+    en = end_sam;
+  endif
+
+  figure(1);
+  clf;
+  plot(s(st:en));
+  axis([1 en-st min(s) max(s)]);
+ 
+  if (nargin == 4)
+
+    % small image
+
+    __gnuplot_set__ terminal png size 420,300
+    ss = sprintf("__gnuplot_set__ output \"%s.png\"", pngname);
+    eval(ss)
+    replot;
+
+    % larger image
+
+    __gnuplot_set__ terminal png size 800,600
+    ss = sprintf("__gnuplot_set__ output \"%s_large.png\"", pngname);
+    eval(ss)
+    replot;
+
+  endif
+
+endfunction

libs/libcodec2/octave/pl2.m

@@ -0,0 +1,50 @@
+% Copyright David Rowe 2009
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+
+function pl2(samname1, samname2, start_sam, end_sam, pngname)
+  
+  fs1=fopen(samname1,"rb");
+  s1=fread(fs1,Inf,"short");
+  fs2=fopen(samname2,"rb");
+  s2=fread(fs2,Inf,"short");
+
+  st = 1;
+  en = length(s1);
+  if (nargin >= 3)
+    st = start_sam;
+  endif
+  if (nargin >= 4)
+    en = end_sam;
+  endif
+
+  figure(1);
+  clf;
+  subplot(211);
+  l1 = strcat("r;",samname1,";");
+  plot(s1(st:en), l1);
+  axis([1 en-st min(s1(st:en)) max(s1(st:en))]);
+  subplot(212);
+  l2 = strcat("r;",samname2,";");
+  plot(s2(st:en),l2);
+  axis([1 en-st min(s1(st:en)) max(s1(st:en))]);
+ 
+  if (nargin == 5)
+
+    % small image
+
+    __gnuplot_set__ terminal png size 420,300
+    s = sprintf("__gnuplot_set__ output \"%s.png\"", pngname);
+    eval(s)
+    replot;
+
+    % larger image
+
+    __gnuplot_set__ terminal png size 800,600
+    s = sprintf("__gnuplot_set__ output \"%s_large.png\"", pngname);
+    eval(s)
+    replot;
+
+  endif
+
+endfunction

libs/libcodec2/octave/plamp.m

@@ -0,0 +1,166 @@
+% Copyright David Rowe 2009
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% Plot ampltiude modelling information from dump files.
+
+function plamp(samname, f)
+  
+  sn_name = strcat(samname,"_sn.txt");
+  Sn = load(sn_name);
+
+  sw_name = strcat(samname,"_sw.txt");
+  Sw = load(sw_name);
+
+  sw__name = strcat(samname,"_sw_.txt");
+  if (file_in_path(".",sw__name))
+    Sw_ = load(sw__name);
+  endif
+
+  model_name = strcat(samname,"_model.txt");
+  model = load(model_name);
+
+  modelq_name = strcat(samname,"_qmodel.txt");
+  if (file_in_path(".",modelq_name))
+    modelq = load(modelq_name);
+  endif
+
+  pw_name = strcat(samname,"_pw.txt");
+  if (file_in_path(".",pw_name))
+    Pw = load(pw_name);
+  endif
+
+  lsp_name = strcat(samname,"_lsp.txt");
+  if (file_in_path(".",lsp_name))
+    lsp = load(lsp_name);
+  endif
+
+  phase_name = strcat(samname,"_phase.txt");
+  if (file_in_path(".",phase_name))
+    phase = load(phase_name);
+  endif
+
+  phase_name_ = strcat(samname,"_phase_.txt");
+  if (file_in_path(".",phase_name_))
+    phase_ = load(phase_name_);
+  endif
+
+  snr_name = strcat(samname,"_snr.txt");
+  if (file_in_path(".",snr_name))
+    snr = load(snr_name);
+  endif
+
+  k = ' ';
+  do 
+    figure(1);
+    clf;
+%    s = [ Sn(2*(f-2)-1,:) Sn(2*(f-2),:) ];
+    s = [ Sn(2*f-1,:) Sn(2*f,:) ];
+    plot(s);
+    axis([1 length(s) -20000 20000]);
+
+    figure(2);
+    Wo = model(f,1);
+    L = model(f,2);
+    Am = model(f,3:(L+2));
+    plot((1:L)*Wo*4000/pi, 20*log10(Am),";Am;r");
+    axis([1 4000 -10 80]);
+    hold on;
+%    plot((0:255)*4000/256, Sw(f-2,:),";Sw;");
+    plot((0:255)*4000/256, Sw(f,:),";Sw;");
+
+    if (file_in_path(".",modelq_name))
+      Amq = modelq(f,3:(L+2));
+      plot((1:L)*Wo*4000/pi, 20*log10(Amq),";Amq;g" );
+      if (file_in_path(".",pw_name))
+        plot((0:255)*4000/256, 10*log10(Pw(f,:)),";Pw;c");
+      endif	
+      signal = Am * Am';
+      noise = (Am-Amq) * (Am-Amq)'; 
+      snr1 = 10*log10(signal/noise);
+      Am_err_label = sprintf(";Am error SNR %4.2f dB;m",snr1);
+      plot((1:L)*Wo*4000/pi, 20*log10(Amq) - 20*log10(Am), Am_err_label);
+    endif
+
+    if (file_in_path(".",snr_name))
+      snr_label = sprintf(";phase SNR %4.2f dB;",snr(f));
+      plot(1,1,snr_label);
+    endif
+
+    % phase model - determine SNR and error spectrum for phase model 1
+
+    if (file_in_path(".",phase_name_))
+      orig  = Am.*exp(j*phase(f,1:L));
+      synth = Am.*exp(j*phase_(f,1:L));
+      signal = orig * orig';
+      noise = (orig-synth) * (orig-synth)';
+      snr_phase = 10*log10(signal/noise);
+
+      phase_err_label = sprintf(";phase_err SNR %4.2f dB;",snr_phase);
+      plot((1:L)*Wo*4000/pi, 20*log10(orig-synth), phase_err_label);
+    endif
+
+    if (file_in_path(".",lsp_name))
+      for l=1:10
+        plot([lsp(f,l)*4000/pi lsp(f,l)*4000/pi], [60 80], 'r');
+      endfor
+    endif
+
+    hold off;
+
+    if (file_in_path(".",phase_name))
+      figure(3);
+      plot((1:L)*Wo*4000/pi, phase(f,1:L), ";phase;");
+      axis;
+      if (file_in_path(".",phase_name_))
+        hold on;
+        plot((1:L)*Wo*4000/pi, phase_(f,1:L), ";phase_;");
+	hold off;
+      endif
+      figure(2);
+    endif
+
+    % autocorrelation function to research voicing est
+    
+    %M = length(s);
+    %sw = s .* hanning(M)';
+    %for k=0:159
+    %  R(k+1) = sw(1:320-k) * sw(1+k:320)';
+    %endfor
+    %figure(4);
+    %R_label = sprintf(";R(k) %3.2f;",max(R(20:159))/R(1));
+    %plot(R/R(1),R_label);
+    %grid
+
+    % interactive menu
+
+    printf("\rframe: %d  menu: n-next  b-back  p-png  q-quit ", f);
+    fflush(stdout);
+    k = kbhit();
+    if (k == 'n')
+      f = f + 1;
+    endif
+    if (k == 'b')
+      f = f - 1;
+    endif
+
+    % optional print to PNG
+
+    if (k == 'p')
+      figure(1);
+      pngname = sprintf("%s_%d_sn.png",samname,f);
+      print(pngname, '-dpng', "-S500,500")
+      pngname = sprintf("%s_%d_sn_large.png",samname,f);
+      print(pngname, '-dpng', "-S800,600")
+
+      figure(2);
+      pngname = sprintf("%s_%d_sw.png",samname,f);
+      print(pngname, '-dpng', "-S500,500")
+      pngname = sprintf("%s_%d_sw_large.png",samname,f);
+      print(pngname, '-dpng', "-S800,600")
+    endif
+
+  until (k == 'q')
+  printf("\n");
+
+endfunction

libs/libcodec2/octave/plinterp.m

@@ -0,0 +1,11 @@
+load ../unittest/tinterp_prev.txt;
+load ../unittest/tinterp_interp.txt;
+load ../unittest/tinterp_next.txt;
+
+clf;
+plot(tinterp_prev(:,1), 20.0*log10(tinterp_prev(:,2)),";prev;")
+hold on;
+plot(tinterp_interp(:,1), 20.0*log10(tinterp_interp(:,2)),'g+-;interp;')
+plot(tinterp_next(:,1), 20.0*log10(tinterp_next(:,2)),'ro-;next;')
+hold off;
+axis([0 pi 0 80])

libs/libcodec2/octave/plnlp.m

@@ -0,0 +1,134 @@
+% Copyright David Rowe 2009
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% Plot NLP states from dump files.
+
+function plnlp(samname, f)
+  
+  sn_name = strcat(samname,"_sn.txt");
+  Sn = load(sn_name);
+
+  sw_name = strcat(samname,"_sw.txt");
+  Sw = load(sw_name);
+
+  fw_name = strcat(samname,"_fw.txt");
+  if (file_in_path(".",fw_name))
+    fw = load(fw_name);
+  endif
+
+  e_name = strcat(samname,"_e.txt");
+  if (file_in_path(".",e_name))
+    e = load(e_name);
+  endif
+
+  p_name = strcat(samname,".p");
+  if (file_in_path(".",p_name))
+    p = load(p_name);
+  endif
+
+  sq_name = strcat(samname,"_sq.txt");
+  if (file_in_path(".",sq_name))
+    sq = load(sq_name);
+  endif
+
+  dec_name = strcat(samname,"_dec.txt");
+  if (file_in_path(".",dec_name))
+    dec = load(dec_name);
+  endif
+
+  do 
+    figure(1);
+    clf;
+    s = [ Sn(2*f-1,:) Sn(2*f,:) ];
+    plot(s, ";Sn;");
+    grid
+    axis([1 length(s) -20000 20000]);
+
+    figure(2);
+    plot((0:255)*4000/256, Sw(f,:),";Sw;");
+    grid
+    axis([1 4000 -10 80]);
+    hold on;
+
+    f0 = 8000/p(f);
+    Wo = 2*pi/p(f);
+    L = floor(pi/Wo);
+    f0_label = sprintf("b;P=%3.1f F0=%3.0f;",p(f),f0);
+    for m=1:L-1
+      plot([ m*Wo*4000/pi m*Wo*4000/pi], [10 60], 'b');
+    endfor
+    plot([ L*Wo*4000/pi L*Wo*4000/pi], [10 60], f0_label);
+
+    hold off;
+
+    if (file_in_path(".",fw_name))
+      figure(3);
+      if (file_in_path(".",e_name))
+         subplot(211);
+      endif
+      plot((0:255)*800/256, fw(f,:)/max(fw(f,:)), ";Fw;");
+      axis([1 400 0 1]);
+      if (file_in_path(".",e_name))
+        subplot(212);
+        e_concat = [ e(2*f-1,:) e(2*f,:) ];
+        plot(e_concat(1:400)/max(e_concat(1:400)), "+;MBE E(f);");
+        axis([1 400 0 1]);
+      endif
+    endif
+
+    if (file_in_path(".",sq_name))
+      figure(4);
+      sq_concat = [ sq(2*f-1,:) sq(2*f,:) ];
+      axis
+      plot(sq_concat, ";sq;");
+    endif
+
+    if (file_in_path(".",dec_name))
+      figure(5);
+      plot(dec(f,:), ";dec;");
+    endif
+    
+    figure(2);
+
+    % interactive menu
+
+    printf("\rframe: %d  menu: n-next  b-back  p-png  q-quit ", f);
+    fflush(stdout);
+    k = kbhit();
+    if (k == 'n')
+      f = f + 1;
+    endif
+    if (k == 'b')
+      f = f - 1;
+    endif
+
+    % optional print to PNG
+
+    if (k == 'p')
+    
+      pngname = sprintf("%s_%d",samname,f);
+
+      % small image
+
+      __gnuplot_set__ terminal png size 420,300
+      ss = sprintf("__gnuplot_set__ output \"%s.png\"", pngname);
+      eval(ss)
+      replot;
+
+      % larger image
+
+      __gnuplot_set__ terminal png size 800,600
+      ss = sprintf("__gnuplot_set__ output \"%s_large.png\"", pngname);
+      eval(ss)
+      replot;
+
+      % for some reason I need this to stop large plot getting wiped
+      __gnuplot_set__ output "/dev/null"
+
+    endif
+
+  until (k == 'q')
+  printf("\n");
+
+endfunction

libs/libcodec2/octave/plphase.m

@@ -0,0 +1,198 @@
+% Copyright David Rowe 2009
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% Plot phase modelling information from dump files.
+
+function plphase(samname, f)
+  
+  sn_name = strcat(samname,"_sn.txt");
+  Sn = load(sn_name);
+
+  sw_name = strcat(samname,"_sw.txt");
+  Sw = load(sw_name);
+
+  model_name = strcat(samname,"_model.txt");
+  model = load(model_name);
+
+  sw__name = strcat(samname,"_sw_.txt");
+  if (file_in_path(".",sw__name))
+    Sw_ = load(sw__name);
+  endif
+
+  pw_name = strcat(samname,"_pw.txt");
+  if (file_in_path(".",pw_name))
+    Pw = load(pw_name);
+  endif
+
+  ak_name = strcat(samname,"_ak.txt");
+  if (file_in_path(".",ak_name))
+    ak = load(ak_name);
+  endif
+
+  phase_name = strcat(samname,"_phase.txt");
+  if (file_in_path(".",phase_name))
+    phase = load(phase_name);
+  endif
+
+  phase_name_ = strcat(samname,"_phase_.txt");
+  if (file_in_path(".",phase_name_))
+    phase_ = load(phase_name_);
+  endif
+
+  snr_name = strcat(samname,"_snr.txt");
+  if (file_in_path(".",snr_name))
+    snr = load(snr_name);
+  endif
+
+  sn_name_ = strcat(samname,".raw");
+  if (file_in_path(".",sn_name_))
+    fs_ = fopen(sn_name_,"rb");
+    sn_  = fread(fs_,Inf,"short");
+  endif
+
+  k = ' ';
+  do 
+    figure(1);
+    clf;
+    s = [ Sn(2*f-1,:) Sn(2*f,:) ];
+    plot(s);
+    grid;
+    axis([1 length(s) -20000 20000]);
+    if (k == 'p')
+       pngname = sprintf("%s_%d_sn",samname,f);
+       png(pngname);
+    endif
+
+    figure(2);
+    Wo = model(f,1);
+    L = model(f,2);
+    Am = model(f,3:(L+2));
+    plot((1:L)*Wo*4000/pi, 20*log10(Am),"r;Am;");
+    axis([1 4000 -10 80]);
+    hold on;
+    plot((0:255)*4000/256, Sw(f,:),";Sw;");
+    grid;
+
+    if (file_in_path(".",sw__name))
+      plot((0:255)*4000/256, Sw_(f,:),"g;Sw_;");
+    endif	
+
+    if (file_in_path(".",pw_name))
+       plot((0:255)*4000/256, 10*log10(Pw(f,:)),";Pw;");
+    endif	
+
+    if (file_in_path(".",snr_name))
+      snr_label = sprintf(";phase SNR %4.2f dB;",snr(f));
+      plot(1,1,snr_label);
+    endif
+
+    % phase model - determine SNR and error spectrum for phase model 1
+
+    if (file_in_path(".",phase_name_))
+      orig  = Am.*exp(j*phase(f,1:L));
+      synth = Am.*exp(j*phase_(f,1:L));
+      signal = orig * orig';
+      noise = (orig-synth) * (orig-synth)';
+      snr_phase = 10*log10(signal/noise);
+
+      phase_err_label = sprintf("g;phase_err SNR %4.2f dB;",snr_phase);
+      plot((1:L)*Wo*4000/pi, 20*log10(orig-synth), phase_err_label);
+    endif
+
+    hold off;
+    if (k == 'p')
+       pngname = sprintf("%s_%d_sw",samname,f);
+       png(pngname);
+    endif
+
+    if (file_in_path(".",phase_name))
+      figure(3);
+      plot((1:L)*Wo*4000/pi, phase(f,1:L)*180/pi, "-o;phase;");
+      axis;
+      if (file_in_path(".", phase_name_))
+        hold on;
+        plot((1:L)*Wo*4000/pi, phase_(f,1:L)*180/pi, "g;phase_;");
+	grid
+	hold off;
+      endif
+      if (k == 'p')
+        pngname = sprintf("%s_%d_phase",samname,f);
+        png(pngname);
+      endif
+    endif
+
+    % synthesised speech 
+
+    if (file_in_path(".",sn_name_))
+      figure(4);
+      s_ = sn_((f-3)*80+1:(f+1)*80);
+      plot(s_);
+      axis([1 length(s_) -20000 20000]);
+      if (k == 'p')
+        pngname = sprintf("%s_%d_sn_",samname,f)
+        png(pngname);
+      endif
+    endif
+
+    if (file_in_path(".",ak_name))
+      figure(5);
+      axis;
+      akw = ak(f,:);
+      weight = 1.0 .^ (0:length(akw)-1);
+      akw = akw .* weight;
+      H = 1./fft(akw,8000);
+      subplot(211);
+      plot(20*log10(abs(H(1:4000))),";LPC mag spec;");
+      grid;	
+      subplot(212);
+      plot(angle(H(1:4000))*180/pi,";LPC phase spec;");
+      grid;
+      if (k == 'p')
+        % stops multimode errors from gnuplot, I know not why...
+        figure(2);
+        figure(5);
+
+        pngname = sprintf("%s_%d_lpc",samname,f);
+        png(pngname);
+      endif
+    endif
+
+
+    % autocorrelation function to research voicing est
+    
+    %M = length(s);
+    %sw = s .* hanning(M)';
+    %for k=0:159
+    %  R(k+1) = sw(1:320-k) * sw(1+k:320)';
+    %endfor
+    %figure(4);
+    %R_label = sprintf(";R(k) %3.2f;",max(R(20:159))/R(1));
+    %plot(R/R(1),R_label);
+    %grid
+
+    figure(2);
+
+    % interactive menu
+
+    printf("\rframe: %d  menu: n-next  b-back  p-png  q-quit ", f);
+    fflush(stdout);
+    k = kbhit();
+    if (k == 'n')
+      f = f + 1;
+    endif
+    if (k == 'b')
+      f = f - 1;
+    endif
+
+    % optional print to PNG
+
+    if (k == 'p')
+       pngname = sprintf("%s_%d",samname,f);
+       png(pngname);
+    endif
+
+  until (k == 'q')
+  printf("\n");
+
+endfunction

libs/libcodec2/octave/plpitch.m

@@ -0,0 +1,36 @@
+% Copyright David Rowe 2009
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% plpitch.m
+% Plots two pitch tracks on top of each other, used for comparing pitch
+% estimators
+
+function plpitch(pitch1_name, pitch2_name, start_fr, end_fr)
+  
+  pitch1 = load(pitch1_name);
+  pitch2 = load(pitch2_name);
+
+  st = 1;
+  en = length(pitch1);
+  if (nargin >= 3)
+    st = start_fr;
+  endif
+  if (nargin >= 4)
+    en = end_fr;
+  endif
+
+  figure(1);
+  clf;
+  l1 = strcat("r;",pitch1_name,";")
+  l1 
+  st
+  en
+  plot(pitch1(st:en), l1);
+  axis([1 en-st 20 160]);
+  l2 = strcat("g;",pitch2_name,";");
+  hold on;
+  plot(pitch2(st:en),l2);
+  hold off;
+endfunction
+

libs/libcodec2/octave/png.m

@@ -0,0 +1,25 @@
+% Copyright David Rowe 2009
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% Replot current plot as a png, generates small and large versions
+
+function png(pngname)
+     % small image
+
+     __gnuplot_set__ terminal png size 420,300
+     ss = sprintf("__gnuplot_set__ output \"%s.png\"", pngname);
+     eval(ss)
+     replot;
+
+     % larger image
+
+     __gnuplot_set__ terminal png size 800,600
+     ss = sprintf("__gnuplot_set__ output \"%s_large.png\"", pngname);
+     eval(ss)
+     replot;
+
+     % for some reason I need this to stop large plot getting wiped
+     __gnuplot_set__ output "/dev/null"
+
+endfunction

libs/libcodec2/octave/postfilter.m

@@ -0,0 +1,24 @@
+% Copyright David Rowe 2009
+% This program is distributed under the terms of the GNU General Public License 
+% Version 2
+%
+% Plot postfilter doing its thing
+
+function postfilter(samname)	 
+    p = load(samname);
+    figure(1);
+    plot(p(:,1),";energy;");
+    hold on;
+    plot(p(:,2),";bg_est;");
+    hold off;
+    grid;
+    pngname=sprintf("%s_postfilter_1", samname);
+    png(pngname);
+
+    figure(2);
+    plot(p(:,3),";% unvoiced;");
+    grid;    
+    pngname=sprintf("%s_postfilter_2", samname);
+    png(pngname);
+endfunction
+  

libs/libcodec2/octave/pulse.m

@@ -0,0 +1,37 @@
+% pulse.m
+% David Rowe August 2009
+%
+% Experiments with human pulse perception for sinusoidal codecs
+
+function pulse(samname)
+
+  A = 1000;
+  K = 16000;
+  N = 80;
+  frames = K/N;
+  s = zeros(1,K);
+
+  for f=1:frames
+    % lets try placing np random pulses in every frame
+
+    P = 20 + (160-20)*rand(1,1);
+    Wo = 2*pi/P;
+    L = floor(pi/Wo);
+    sf = zeros(1,N);
+    for m=1:L/2:L
+      pos = floor(rand(1,1)*N)+1;
+      %pos = 50;
+      for l=m:m+L/2-1
+        sf = sf + A*cos(l*Wo*((f-1)*N+1:f*N) - pos*l*Wo);
+      endfor
+    endfor
+    s((f-1)*N+1:f*N) = sf;
+  endfor
+
+  plot(s(1:250));
+
+  fs=fopen(samname,"wb");
+  fwrite(fs,s,"short");
+  fclose(fs);
+endfunction
+

libs/libcodec2/pitch/hts1a.p

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libs/libcodec2/pitch/hts2a.p

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libs/libcodec2/script/menu.sh

@@ -0,0 +1,70 @@
+#!/bin/bash
+# ./menu.sh
+#
+# David Rowe
+# Created August 2009
+#
+# Presents a menu of sound files, press 1 to play file1, 2 to play file2 etc
+#
+# The aim is to make comparing files with different processing easier than
+# using up-arrow on the command line.  Based on cdialog.
+#
+# usage:
+#   menu.sh file1.raw file2.raw ........ [-d playbackdevice]
+#
+# for example:
+#
+#   ../script/menu.sh hts1a.raw hts1a_uq.raw 
+#
+# or:
+#
+#   ../script/menu.sh hts1a.raw hts1a_uq.raw -d /dev/dsp1
+#
+
+#  Copyright (C) 2007 David Rowe
+# 
+#  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.
+
+files=0
+items="Q-Quit\n"
+while [ ! -z "$1" ]
+do
+  case "$1" in
+    -d) dsp="${1} ${2}"; shift;;
+     *) files=`expr 1 + $files`;
+        new_file=$1;
+        file[$files]=$new_file;
+        items="${items} ${files}-${new_file}\n";;
+  esac
+  shift
+done
+
+readchar=1
+echo -n -e "\r" $items"- "
+while [ $readchar -ne 0 ]
+do
+  echo -n -e "\r -"
+  stty cbreak         # or stty raw
+  readchar=`dd if=/dev/tty bs=1 count=1 2>/dev/null`
+  stty -cbreak
+  if [ $readchar == 'q' ] ; then
+    readchar=0
+  fi
+  if [ $readchar -ne 0 ] ; then
+    play -r 8000 -s -2 ${file[$readchar]} $dsp 2> /dev/null
+  fi
+done
+echo

libs/libcodec2/script/playraw.sh

@@ -0,0 +1,6 @@
+#!/bin/sh
+# Plays a raw file
+# usage:
+#   playraw file.raw
+#   playraw file.raw -d /dev/dsp1 (e.g. for USB headphones)
+play -r 8000 -s -2 $1 $2 $3

libs/libcodec2/script/raw2wav.sh

@@ -0,0 +1,3 @@
+#!/bin/sh
+# Converts 16 bit signed short 8 kHz raw (headerless) files to wave
+sox -r 8000 -s -2 $1 $2

libs/libcodec2/script/wav2raw.sh

@@ -0,0 +1,3 @@
+#!/bin/sh
+# Converts wave files to raw (headerless) files
+sox $1 -t raw $2

libs/libcodec2/src/Makefile.am

@@ -0,0 +1,53 @@
+AM_CFLAGS = -I../src -Wall -DFLOATING_POINT -DVAR_ARRAYS
+AUTOMAKE_OPTS = gnu
+NAME = libcodec2
+AM_CPPFLAGS = $(AM_CFLAGS)
+
+lib_LTLIBRARIES	= libcodec2.la
+libcodec2_la_SOURCES = dump.c \
+lpc.c \
+nlp.c \
+postfilter.c \
+sine.c \
+codec2.c \
+four1.c \
+interp.c \
+lsp.c \
+phase.c \
+quantise.c \
+pack.c \
+codebook.c
+
+libcodec2_la_CFLAGS = $(AM_CFLAGS)
+libcodec2_la_LDFLAGS = $(LIBS)
+
+library_includedir = $(prefix)
+library_include_HEADERS	= codec2.h \
+defines.h \
+four1.h \
+interp.h \
+lsp.h \
+phase.h \
+quantise.h \
+comp.h \
+dump.h \
+globals.h \
+lpc.h \
+nlp.h  \
+postfilter.h \
+sine.h \
+codebook.h
+
+bin_PROGRAMS = c2dec c2enc c2sim
+
+c2dec_SOURCES = c2dec.c
+c2dec_LDADD = $(lib_LTLIBRARIES) 
+c2dec_LDFLAGS = $(LIBS)
+
+c2enc_SOURCES = c2enc.c
+c2enc_LDADD = $(lib_LTLIBRARIES) 
+c2enc_LDFLAGS = $(LIBS)
+
+c2sim_SOURCES = c2sim.c
+c2sim_LDADD = $(lib_LTLIBRARIES) 
+c2sim_LDFLAGS = $(LIBS)

libs/libcodec2/src/Makefile.in

@@ -0,0 +1,690 @@
+# Makefile.in generated by automake 1.9.6 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002,
+# 2003, 2004, 2005  Free Software Foundation, Inc.
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
+@SET_MAKE@
+
+
+
+srcdir = @srcdir@
+top_srcdir = @top_srcdir@
+VPATH = @srcdir@
+pkgdatadir = $(datadir)/@PACKAGE@
+pkglibdir = $(libdir)/@PACKAGE@
+pkgincludedir = $(includedir)/@PACKAGE@
+top_builddir = ..
+am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd
+INSTALL = @INSTALL@
+install_sh_DATA = $(install_sh) -c -m 644
+install_sh_PROGRAM = $(install_sh) -c
+install_sh_SCRIPT = $(install_sh) -c
+INSTALL_HEADER = $(INSTALL_DATA)
+transform = $(program_transform_name)
+NORMAL_INSTALL = :
+PRE_INSTALL = :
+POST_INSTALL = :
+NORMAL_UNINSTALL = :
+PRE_UNINSTALL = :
+POST_UNINSTALL = :
+build_triplet = @build@
+host_triplet = @host@
+bin_PROGRAMS = c2dec$(EXEEXT) c2enc$(EXEEXT) c2sim$(EXEEXT)
+subdir = src
+DIST_COMMON = $(library_include_HEADERS) $(srcdir)/Makefile.am \
+	$(srcdir)/Makefile.in
+ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
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+
+libcodec2_la-codec2.lo: codec2.c
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+@am__fastdepCC_TRUE@	then mv -f "$(DEPDIR)/libcodec2_la-codec2.Tpo" "$(DEPDIR)/libcodec2_la-codec2.Plo"; else rm -f "$(DEPDIR)/libcodec2_la-codec2.Tpo"; exit 1; fi
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	source='codec2.c' object='libcodec2_la-codec2.lo' libtool=yes @AMDEPBACKSLASH@
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+
+libcodec2_la-four1.lo: four1.c
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+@am__fastdepCC_TRUE@	then mv -f "$(DEPDIR)/libcodec2_la-four1.Tpo" "$(DEPDIR)/libcodec2_la-four1.Plo"; else rm -f "$(DEPDIR)/libcodec2_la-four1.Tpo"; exit 1; fi
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	source='four1.c' object='libcodec2_la-four1.lo' libtool=yes @AMDEPBACKSLASH@
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+
+libcodec2_la-interp.lo: interp.c
+@am__fastdepCC_TRUE@	if $(LIBTOOL) --tag=CC --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libcodec2_la_CFLAGS) $(CFLAGS) -MT libcodec2_la-interp.lo -MD -MP -MF "$(DEPDIR)/libcodec2_la-interp.Tpo" -c -o libcodec2_la-interp.lo `test -f 'interp.c' || echo '$(srcdir)/'`interp.c; \
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+
+libcodec2_la-lsp.lo: lsp.c
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+
+libcodec2_la-phase.lo: phase.c
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+
+libcodec2_la-quantise.lo: quantise.c
+@am__fastdepCC_TRUE@	if $(LIBTOOL) --tag=CC --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libcodec2_la_CFLAGS) $(CFLAGS) -MT libcodec2_la-quantise.lo -MD -MP -MF "$(DEPDIR)/libcodec2_la-quantise.Tpo" -c -o libcodec2_la-quantise.lo `test -f 'quantise.c' || echo '$(srcdir)/'`quantise.c; \
+@am__fastdepCC_TRUE@	then mv -f "$(DEPDIR)/libcodec2_la-quantise.Tpo" "$(DEPDIR)/libcodec2_la-quantise.Plo"; else rm -f "$(DEPDIR)/libcodec2_la-quantise.Tpo"; exit 1; fi
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	source='quantise.c' object='libcodec2_la-quantise.lo' libtool=yes @AMDEPBACKSLASH@
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+
+libcodec2_la-pack.lo: pack.c
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+@am__fastdepCC_TRUE@	then mv -f "$(DEPDIR)/libcodec2_la-pack.Tpo" "$(DEPDIR)/libcodec2_la-pack.Plo"; else rm -f "$(DEPDIR)/libcodec2_la-pack.Tpo"; exit 1; fi
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+
+libcodec2_la-codebook.lo: codebook.c
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+@am__fastdepCC_TRUE@	then mv -f "$(DEPDIR)/libcodec2_la-codebook.Tpo" "$(DEPDIR)/libcodec2_la-codebook.Plo"; else rm -f "$(DEPDIR)/libcodec2_la-codebook.Tpo"; exit 1; fi
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+
+mostlyclean-libtool:
+	-rm -f *.lo
+
+clean-libtool:
+	-rm -rf .libs _libs
+
+distclean-libtool:
+	-rm -f libtool
+uninstall-info-am:
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+	  if test -f "$$p"; then d=; else d="$(srcdir)/"; fi; \
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+	done
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+	  echo " rm -f '$(DESTDIR)$(library_includedir)/$$f'"; \
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+	    echo "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'"` install
+mostlyclean-generic:
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+clean-generic:
+
+distclean-generic:
+	-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+
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+	@echo "it deletes files that may require special tools to rebuild."
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libs/libcodec2/src/c2dec.c

@@ -0,0 +1,80 @@
+/*---------------------------------------------------------------------------*\
+
+  FILE........: c2dec.c
+  AUTHOR......: David Rowe
+  DATE CREATED: 23/8/2010
+
+  Decodes a file of bits to a file of raw speech samples using codec2. Demo
+  program for codec2.
+
+  NOTE: the bit file is not packed, 51 bits/frame actually consumes 51
+  bytes/frame on disk.  If you are using this for a real world
+  application you may want to pack the 51 bytes into 7 bytes.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2010 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#include "codec2.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+
+int main(int argc, char *argv[])
+{
+    static const int bitsSize = ((CODEC2_BITS_PER_FRAME + 7) / 8);
+    void *codec2;
+    FILE *fin;
+    FILE *fout;
+    short buf[CODEC2_SAMPLES_PER_FRAME];
+    unsigned char  bits[bitsSize];
+
+    if (argc != 3) {
+	printf("usage: %s InputBitFile OutputRawSpeechFile\n", argv[0]);
+	exit(1);
+    }
+ 
+    if ( (fin = fopen(argv[1],"rb")) == NULL ) {
+	fprintf(stderr, "Error opening input bit file: %s: %s.\n",
+         argv[1], strerror(errno));
+	exit(1);
+    }
+
+    if ( (fout = fopen(argv[2],"wb")) == NULL ) {
+	fprintf(stderr, "Error opening output speech file: %s: %s.\n",
+         argv[2], strerror(errno));
+	exit(1);
+    }
+
+    codec2 = codec2_create();
+
+    while(fread(bits, sizeof(char), bitsSize, fin) == bitsSize) {
+	codec2_decode(codec2, buf, bits);
+	fwrite(buf, sizeof(short), CODEC2_SAMPLES_PER_FRAME, fout);
+    }
+
+    codec2_destroy(codec2);
+
+    fclose(fin);
+    fclose(fout);
+
+    return 0;
+}

libs/libcodec2/src/c2enc.c

@@ -0,0 +1,82 @@
+/*---------------------------------------------------------------------------*\
+
+  FILE........: c2enc.c
+  AUTHOR......: David Rowe
+  DATE CREATED: 23/8/2010
+
+  Encodes a file of raw speech samples using codec2 and ouputs a file
+  of bits (each bit is stored in the LSB or each output byte). Demo
+  program for codec2.
+
+  NOTE: the bit file is not packed, 51 bits/frame actually consumes 51
+  bytes/frame on disk.  If you are using this for a real world
+  application you may want to pack the 51 bytes into 7 bytes.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2010 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#include "codec2.h"
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+
+int main(int argc, char *argv[])
+{
+    static const int bitsSize = ((CODEC2_BITS_PER_FRAME + 7) / 8);
+    void *codec2;
+    FILE *fin;
+    FILE *fout;
+    short buf[CODEC2_SAMPLES_PER_FRAME];
+    unsigned char  bits[bitsSize];
+
+    if (argc != 3) {
+	printf("usage: %s InputRawspeechFile OutputBitFile\n", argv[0]);
+	exit(1);
+    }
+ 
+    if ( (fin = fopen(argv[1],"rb")) == NULL ) {
+	fprintf(stderr, "Error opening input bit file: %s: %s.\n",
+         argv[1], strerror(errno));
+	exit(1);
+    }
+
+    if ( (fout = fopen(argv[2],"wb")) == NULL ) {
+	fprintf(stderr, "Error opening output speech file: %s: %s.\n",
+         argv[2], strerror(errno));
+	exit(1);
+    }
+
+    codec2 = codec2_create();
+
+    while(fread(buf, sizeof(short), CODEC2_SAMPLES_PER_FRAME, fin) ==
+	  CODEC2_SAMPLES_PER_FRAME) {
+	codec2_encode(codec2, bits, buf);
+	fwrite(bits, sizeof(char), bitsSize, fout);
+    }
+
+    codec2_destroy(codec2);
+
+    fclose(fin);
+    fclose(fout);
+
+    return 0;
+}

libs/libcodec2/src/c2sim.c

@@ -0,0 +1,408 @@
+/*---------------------------------------------------------------------------*\
+
+  FILE........: c2sim.c
+  AUTHOR......: David Rowe
+  DATE CREATED: 20/8/2010
+
+  Codec2 simulation.  Combines encoder and decoder and allows switching in 
+  out various algorithms and quantisation steps. 
+
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#include <assert.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <errno.h>
+#include <math.h>
+
+#include "defines.h"
+#include "sine.h"
+#include "nlp.h"
+#include "dump.h"
+#include "lpc.h"
+#include "lsp.h"
+#include "quantise.h"
+#include "phase.h"
+#include "postfilter.h"
+#include "interp.h"
+
+/*---------------------------------------------------------------------------*\
+                                                                             
+ switch_present()                                                            
+                                                                             
+ Searches the command line arguments for a "switch".  If the switch is       
+ found, returns the command line argument where it ws found, else returns    
+ NULL.                                                                       
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+int switch_present(sw,argc,argv)
+register char sw[];     /* switch in string form */
+register int argc;      /* number of command line arguments */
+register char *argv[];  /* array of command line arguments in string form */
+{
+  register int i;       /* loop variable */
+
+  for(i=1; i<argc; i++)
+    if (!strcmp(sw,argv[i]))
+      return(i);
+
+  return 0;
+}
+
+void synth_one_frame(short buf[], MODEL *model, float Sn_[], float Pn[]);
+
+/*---------------------------------------------------------------------------*\
+                                                                          
+				MAIN                                        
+                                                                         
+\*---------------------------------------------------------------------------*/
+
+int main(int argc, char *argv[])
+{
+  FILE *fout;		/* output speech file                    */
+  FILE *fin;		/* input speech file                     */
+  short buf[N];		/* input/output buffer                   */
+  float Sn[M];	        /* float input speech samples            */
+  COMP  Sw[FFT_ENC];	/* DFT of Sn[]                           */
+  float w[M];	        /* time domain hamming window            */
+  COMP  W[FFT_ENC];	/* DFT of w[]                            */
+  MODEL model;
+  float Pn[2*N];	/* trapezoidal synthesis window          */
+  float Sn_[2*N];	/* synthesised speech */
+  int   i;		/* loop variable                         */
+  int   frames;
+  float prev_Wo;
+  float pitch;
+  int   voiced1;
+
+  char  out_file[MAX_STR];
+  int   arg;
+  float snr;
+  float sum_snr;
+
+  int lpc_model, order;
+  int lsp, lsp_quantiser;
+  float ak[LPC_MAX];
+  COMP  Sw_[FFT_ENC];
+  
+  int dump;
+  
+  int phase0;
+  float ex_phase[MAX_AMP+1];
+
+  int   postfilt;
+  float bg_est;
+
+  int   hand_voicing;
+  FILE *fvoicing;
+
+  MODEL prev_model, interp_model;
+  int decimate;
+
+  void *nlp_states;
+
+  for(i=0; i<M; i++)
+      Sn[i] = 1.0;
+  for(i=0; i<2*N; i++)
+      Sn_[i] = 0;
+
+  prev_Wo = TWO_PI/P_MAX;
+
+  prev_model.Wo = TWO_PI/P_MIN;
+  prev_model.L = floor(PI/prev_model.Wo);
+  for(i=1; i<=prev_model.L; i++) {
+      prev_model.A[i] = 0.0;
+      prev_model.phi[i] = 0.0;
+  }
+  for(i=1; i<=MAX_AMP; i++) {
+      ex_phase[i] = 0.0;
+  }
+
+  nlp_states = nlp_create();
+
+  if (argc < 2) {
+    fprintf(stderr, "\nCodec2 - 2400 bit/s speech codec - Simulation Program\n"
+     "\thttp://rowetel.com/codec2.html\n\n"
+     "usage: %s InputFile [-o OutputFile]\n"
+     "\t[-o lpc Order]\n"
+     "\t[--lsp]\n"
+     "\t[--phase0]\n"
+     "\t[--postfilter]\n"
+     "\t[--hand_voicing]\n"
+     "\t[--dec]\n"
+     "\t[--dump DumpFilePrefix]\n", argv[0]);
+    exit(1);
+  }
+
+  /* Interpret command line arguments -------------------------------------*/
+
+  /* Input file */
+
+  if ((fin = fopen(argv[1],"rb")) == NULL) {
+    fprintf(stderr, "Error opening input bit file: %s: %s.\n",
+     argv[1], strerror(errno));
+    exit(1);
+  }
+
+  /* Output file */
+
+  if ((arg = switch_present("-o",argc,argv))) {
+    if ((fout = fopen(argv[arg+1],"wb")) == NULL) {
+      fprintf(stderr, "Error opening output speech file: %s: %s.\n",
+       argv[arg+1], strerror(errno));
+      exit(1);
+    }
+    strcpy(out_file,argv[arg+1]);
+  }
+  else
+    fout = NULL;
+
+  lpc_model = 0;
+  if ((arg = switch_present("--lpc",argc,argv))) {
+      lpc_model = 1;
+      order = atoi(argv[arg+1]);
+      if ((order < 4) || (order > 20)) {
+        fprintf(stderr, "Error in lpc order: %d\n", order);
+        exit(1);
+      }	  
+  }
+
+  dump = switch_present("--dump",argc,argv);
+  if (dump) 
+      dump_on(argv[dump+1]);
+
+  lsp = switch_present("--lsp",argc,argv);
+  lsp_quantiser = 0;
+
+  phase0 = switch_present("--phase0",argc,argv);
+  if (phase0) {
+      ex_phase[0] = 0;
+  }
+
+  hand_voicing = switch_present("--hand_voicing",argc,argv);
+  if (hand_voicing) {
+      fvoicing = fopen(argv[hand_voicing+1],"rt");
+      assert(fvoicing != NULL);
+  }
+
+  bg_est = 0.0;
+  postfilt = switch_present("--postfilter",argc,argv);
+
+  decimate = switch_present("--dec",argc,argv);
+
+  /* Initialise ------------------------------------------------------------*/
+
+  make_analysis_window(w,W);
+  make_synthesis_window(Pn);
+  quantise_init();
+
+  /* Main loop ------------------------------------------------------------*/
+
+  frames = 0;
+  sum_snr = 0;
+  while(fread(buf,sizeof(short),N,fin)) {
+    frames++;
+    
+    /* Read input speech */
+
+    for(i=0; i<M-N; i++)
+      Sn[i] = Sn[i+N];
+    for(i=0; i<N; i++)
+      Sn[i+M-N] = buf[i];
+ 
+    /* Estimate pitch */
+
+    nlp(nlp_states,Sn,N,M,P_MIN,P_MAX,&pitch,Sw,&prev_Wo);
+    prev_Wo = TWO_PI/pitch;
+    model.Wo = TWO_PI/pitch;
+
+    /* estimate model parameters */
+
+    dft_speech(Sw, Sn, w); 
+    two_stage_pitch_refinement(&model, Sw);
+    estimate_amplitudes(&model, Sw, W);
+    dump_Sn(Sn); dump_Sw(Sw); dump_model(&model);
+
+    /* optional zero-phase modelling */
+
+    if (phase0) {
+	float Wn[M];		        /* windowed speech samples */
+	float Rk[LPC_ORD+1];	        /* autocorrelation coeffs  */
+  	
+	dump_phase(&model.phi[0], model.L);
+
+	/* find aks here, these are overwritten if LPC modelling is enabled */
+
+	for(i=0; i<M; i++)
+	    Wn[i] = Sn[i]*w[i];
+	autocorrelate(Wn,Rk,M,LPC_ORD);
+	levinson_durbin(Rk,ak,LPC_ORD);
+
+	if (lpc_model)
+	    assert(order == LPC_ORD);
+
+	dump_ak(ak, LPC_ORD);
+	
+	/* determine voicing */
+
+	snr = est_voicing_mbe(&model, Sw, W, (FS/TWO_PI)*model.Wo, Sw_);
+	dump_Sw_(Sw_);
+	dump_snr(snr);
+
+	/* just to make sure we are not cheating - kill all phases */
+
+	for(i=0; i<MAX_AMP; i++)
+	    model.phi[i] = 0;
+	
+	if (hand_voicing) {
+	    fscanf(fvoicing,"%d\n",&model.voiced);
+	}
+    }
+ 
+    /* optional LPC model amplitudes */
+
+    if (lpc_model) {
+	int lpc_correction;
+	float e;
+	float lsps[LPC_ORD];
+	int   lsp_indexes[LPC_ORD];
+
+	e = speech_to_uq_lsps(lsps, ak, Sn, w, order);
+	lpc_correction = need_lpc_correction(&model, ak, e);
+
+	if (lsp) {
+	    encode_lsps(lsp_indexes, lsps, LPC_ORD);
+	    /*
+	      for(i=0; i<LPC_ORD; i++)
+		printf("lsps[%d] = %f lsp_indexes[%d] = %d\n", 
+		       i, lsps[i], i, lsp_indexes[i]);
+	      printf("\n");
+	    */
+	    decode_lsps(lsps, lsp_indexes, LPC_ORD);
+	    bw_expand_lsps(lsps, LPC_ORD);
+	    lsp_to_lpc(lsps, ak, LPC_ORD);
+	}
+
+	e = decode_energy(encode_energy(e));
+	model.Wo = decode_Wo(encode_Wo(model.Wo));
+
+	aks_to_M2(ak, order, &model, e, &snr, 1); 
+	apply_lpc_correction(&model, lpc_correction);
+	sum_snr += snr;
+        dump_quantised_model(&model);
+    }
+
+    /* option decimation to 20ms rate, which enables interpolation
+       routine to synthesise in between frame */
+  
+    if (decimate) {
+	if (!phase0) {
+	    printf("needs --phase0 to resample phase for interpolated Wo\n");
+	    exit(0);
+	}
+
+	/* odd frame - interpolate */
+
+	if (frames%2) {
+
+	    #ifdef TEST
+	    model.voiced = 1;
+	    prev_model.voiced = 1;
+	    if (fabs(prev_model.Wo - model.Wo) < 0.1*model.Wo) {
+		interp_model.voiced = 1;
+		interpolate(&interp_model, &prev_model, &model);
+		for(i=0; i<=interp_model.L; i++) {
+		    interp_model.phi[i] = phi1[i];
+		}
+		printf("interp\n");
+	    }
+	    else
+		interp_model = tmp_model;
+	    #endif
+
+	    interp_model.voiced = voiced1;
+	    interpolate(&interp_model, &prev_model, &model);
+	    
+	    if (phase0)
+		phase_synth_zero_order(&interp_model, ak, ex_phase);	
+	    if (postfilt)
+		postfilter(&interp_model, &bg_est);
+	    synth_one_frame(buf, &interp_model, Sn_, Pn);
+	    if (fout != NULL) fwrite(buf,sizeof(short),N,fout);
+
+	    if (phase0)
+		phase_synth_zero_order(&model, ak, ex_phase);	
+	    if (postfilt)
+		postfilter(&model, &bg_est);
+	    synth_one_frame(buf, &model, Sn_, Pn);
+	    if (fout != NULL) fwrite(buf,sizeof(short),N,fout);
+
+	    prev_model = model;
+	}
+	else {
+	    voiced1 = model.voiced;
+	}
+    }
+    else {
+	if (phase0)
+	   phase_synth_zero_order(&model, ak, ex_phase);	
+	if (postfilt)
+	    postfilter(&model, &bg_est);
+	synth_one_frame(buf, &model, Sn_, Pn);
+	if (fout != NULL) fwrite(buf,sizeof(short),N,fout);
+    }
+  }
+
+  if (fout != NULL)
+    fclose(fout);
+
+  if (lpc_model)
+      printf("SNR av = %5.2f dB\n", sum_snr/frames);
+
+  if (dump)
+      dump_off();
+
+  if (hand_voicing)
+    fclose(fvoicing);
+
+  nlp_destroy(nlp_states);
+
+  return 0;
+}
+
+void synth_one_frame(short buf[], MODEL *model, float Sn_[], float Pn[])
+{
+    int     i;
+
+    synthesise(Sn_, model, Pn, 1);
+
+    for(i=0; i<N; i++) {
+	if (Sn_[i] > 32767.0)
+	    buf[i] = 32767;
+	else if (Sn_[i] < -32767.0)
+	    buf[i] = -32767;
+	else
+	    buf[i] = Sn_[i];
+    }
+
+}

libs/libcodec2/src/codeall.sh

@@ -0,0 +1,9 @@
+#!/bin/sh
+# codeall.sh
+# David Rowe 24 sep 2009
+# Code all samples using various processing steps
+./code.sh hts1a
+./code.sh hts2a
+./code.sh mmt1
+./code.sh morig
+./code.sh forig

libs/libcodec2/src/codebook.c

@@ -0,0 +1,162 @@
+float codebook_lsp1[] = {
+	225,
+	250,
+	275,
+	300,
+	325,
+	350,
+	375,
+	400,
+	425,
+	450,
+	475,
+	500,
+	525,
+	550,
+	575,
+	600,0,0,0,0,0,0,0,0,0,0
+};
+
+float codebook_lsp2[] = {
+	325,
+	350,
+	375,
+	400,
+	425,
+	450,
+	475,
+	500,
+	525,
+	550,
+	575,
+	600,
+	625,
+	650,
+	675,
+	700,0,0,0,0,0,0,0,0,0,0
+};
+
+float codebook_lsp3[] = {
+	500,
+	550,
+	600,
+	650,
+	700,
+	750,
+	800,
+	850,
+	900,
+	950,
+	1000,
+	1050,
+	1100,
+	1150,
+	1200,
+	1250,0,0,0,0,0,0,0,0,0,0
+};
+
+float codebook_lsp4[] = {
+	700,
+	800,
+	900,
+	1000,
+	1100,
+	1200,
+	1300,
+	1400,
+	1500,
+	1600,
+	1700,
+	1800,
+	1900,
+	2000,
+	2100,
+	2200,0,0,0,0,0,0,0,0,0,0
+};
+
+float codebook_lsp5[] = {
+	950,
+	1050,
+	1150,
+	1250,
+	1350,
+	1450,
+	1550,
+	1650,
+	1750,
+	1850,
+	1950,
+	2050,
+	2150,
+	2250,
+	2350,
+	2450,0,0,0,0,0,0,0,0,0,0
+};
+
+float codebook_lsp6[] = {
+	1100,
+	1200,
+	1300,
+	1400,
+	1500,
+	1600,
+	1700,
+	1800,
+	1900,
+	2000,
+	2100,
+	2200,
+	2300,
+	2400,
+	2500,
+	2600,0,0,0,0,0,0,0,0,0,0
+};
+
+float codebook_lsp7[] = {
+	1500,
+	1600,
+	1700,
+	1800,
+	1900,
+	2000,
+	2100,
+	2200,
+	2300,
+	2400,
+	2500,
+	2600,
+	2700,
+	2800,
+	2900,
+	3000,0,0,0,0,0,0,0,0,0,0
+};
+
+float codebook_lsp8[] = {
+	2300,
+	2400,
+	2500,
+	2600,
+	2700,
+	2800,
+	2900,
+	3000,0,0,0,0,0,0,0,0,0,0
+};
+
+float codebook_lsp9[] = {
+	2500,
+	2600,
+	2700,
+	2800,
+	2900,
+	3000,
+	3100,
+	3200,0,0,0,0,0,0,0,0,0,0
+};
+
+float codebook_lsp10[] = {
+	2900,
+	3100,
+	3300,
+	3500,0,0,0,0,0,0,0,0,0,0
+};
+

libs/libcodec2/src/codebook.h

@@ -0,0 +1,15 @@
+#ifndef CODEBOOK_H
+#define CODEBOOK_H
+
+extern float codebook_lsp1[];
+extern float codebook_lsp2[];
+extern float codebook_lsp3[];
+extern float codebook_lsp4[];
+extern float codebook_lsp5[];
+extern float codebook_lsp6[];
+extern float codebook_lsp7[];
+extern float codebook_lsp8[];
+extern float codebook_lsp9[];
+extern float codebook_lsp10[];
+
+#endif

libs/libcodec2/src/codec2.c

@@ -0,0 +1,337 @@
+/*---------------------------------------------------------------------------*\
+
+  FILE........: codec2.c
+  AUTHOR......: David Rowe
+  DATE CREATED: 21/8/2010
+
+  Codec2 fully quantised encoder and decoder functions.  If you want use 
+  codec2, the codec2_xxx functions are for you.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2010 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#include <assert.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+
+#include "defines.h"
+#include "sine.h"
+#include "nlp.h"
+#include "dump.h"
+#include "lpc.h"
+#include "quantise.h"
+#include "phase.h"
+#include "interp.h"
+#include "postfilter.h"
+#include "codec2.h"
+
+typedef struct {
+    float  Sn[M];        /* input speech                              */
+    float  w[M];	 /* time domain hamming window                */
+    COMP   W[FFT_ENC];	 /* DFT of w[]                                */
+    float  Pn[2*N];	 /* trapezoidal synthesis window              */
+    float  Sn_[2*N];	 /* synthesised speech                        */
+    float  prev_Wo;      /* previous frame's pitch estimate           */
+    float  ex_phase;     /* excitation model phase track              */
+    float  bg_est;       /* background noise estimate for post filter */
+    MODEL  prev_model;   /* model parameters from 20ms ago            */
+    void  *nlp;          /* pitch predictor states                    */
+} CODEC2;
+
+/*---------------------------------------------------------------------------*\
+                                                       
+                             FUNCTION HEADERS
+
+\*---------------------------------------------------------------------------*/
+
+void analyse_one_frame(CODEC2 *c2, MODEL *model, short speech[]);
+void synthesise_one_frame(CODEC2 *c2, short speech[], MODEL *model,float ak[]);
+
+/*---------------------------------------------------------------------------*\
+                                                       
+                                FUNCTIONS
+
+\*---------------------------------------------------------------------------*/
+
+/*---------------------------------------------------------------------------*\
+                                                       
+  FUNCTION....: codec2_create	     
+  AUTHOR......: David Rowe			      
+  DATE CREATED: 21/8/2010 
+
+  Create and initialise an instance of the codec.  Returns a pointer
+  to the codec states or NULL on failure.  One set of states is
+  sufficient for a full duuplex codec (i.e. an encoder and decoder).
+  You don't need separate states for encoders and decoders.  See
+  c2enc.c and c2dec.c for examples.
+
+\*---------------------------------------------------------------------------*/
+
+void *codec2_create()
+{
+    CODEC2 *c2;
+    int     i,l;
+
+    c2 = (CODEC2*)malloc(sizeof(CODEC2));
+    if (c2 == NULL)
+	return NULL;
+
+    for(i=0; i<M; i++)
+	c2->Sn[i] = 1.0;
+    for(i=0; i<2*N; i++)
+	c2->Sn_[i] = 0;
+    make_analysis_window(c2->w,c2->W);
+    make_synthesis_window(c2->Pn);
+    quantise_init();
+    c2->prev_Wo = 0.0;
+    c2->bg_est = 0.0;
+    c2->ex_phase = 0.0;
+
+    for(l=1; l<=MAX_AMP; l++)
+	c2->prev_model.A[l] = 0.0;
+    c2->prev_model.Wo = TWO_PI/P_MAX;
+
+    c2->nlp = nlp_create();
+    if (c2->nlp == NULL) {
+	free (c2);
+	return NULL;
+    }
+
+    return (void*)c2;
+}
+
+/*---------------------------------------------------------------------------*\
+                                                       
+  FUNCTION....: codec2_create	     
+  AUTHOR......: David Rowe			      
+  DATE CREATED: 21/8/2010 
+
+  Destroy an instance of the codec.
+
+\*---------------------------------------------------------------------------*/
+
+void codec2_destroy(void *codec2_state)
+{
+    CODEC2 *c2;
+    
+    assert(codec2_state != NULL);
+    c2 = (CODEC2*)codec2_state;
+    nlp_destroy(c2->nlp);
+    free(codec2_state);
+}
+
+/*---------------------------------------------------------------------------*\
+                                                       
+  FUNCTION....: codec2_encode	     
+  AUTHOR......: David Rowe			      
+  DATE CREATED: 21/8/2010 
+
+  Encodes 160 speech samples (20ms of speech) into 51 bits.  
+
+  The codec2 algorithm actually operates internally on 10ms (80
+  sample) frames, so we run the encoding algorithm twice.  On the
+  first frame we just send the voicing bit.  One the second frame we
+  send all model parameters.
+
+  The bit allocation is:
+
+    Parameter                      bits/frame
+    --------------------------------------
+    Harmonic magnitudes (LSPs)     36
+    Low frequency LPC correction    1
+    Energy                          5
+    Wo (fundamental frequnecy)      7
+    Voicing (10ms update)           2
+    TOTAL                          51
+ 
+\*---------------------------------------------------------------------------*/
+
+void codec2_encode(void *codec2_state, unsigned char * bits, short speech[])
+{
+    CODEC2 *c2;
+    MODEL   model;
+    int     voiced1, voiced2;
+    int     lsp_indexes[LPC_ORD];
+    int     lpc_correction;
+    int     energy_index;
+    int     Wo_index;
+    int     i;
+    unsigned int nbit = 0;
+
+    assert(codec2_state != NULL);
+    c2 = (CODEC2*)codec2_state;
+
+    /* first 10ms analysis frame - we just want voicing */
+
+    analyse_one_frame(c2, &model, speech);
+    voiced1 = model.voiced;
+
+    /* second 10ms analysis frame */
+
+    analyse_one_frame(c2, &model, &speech[N]);
+    voiced2 = model.voiced;
+    
+    Wo_index = encode_Wo(model.Wo);
+    encode_amplitudes(lsp_indexes, 
+		      &lpc_correction, 
+		      &energy_index,
+		      &model, 
+		       c2->Sn, 
+		       c2->w);   
+    memset(bits, '\0', ((CODEC2_BITS_PER_FRAME + 7) / 8));
+    pack(bits, &nbit, Wo_index, WO_BITS);
+    for(i=0; i<LPC_ORD; i++) {
+	pack(bits, &nbit, lsp_indexes[i], lsp_bits(i));
+    }
+    pack(bits, &nbit, lpc_correction, 1);
+    pack(bits, &nbit, energy_index, E_BITS);
+    pack(bits, &nbit, voiced1, 1);
+    pack(bits, &nbit, voiced2, 1);
+    
+    assert(nbit == CODEC2_BITS_PER_FRAME);
+}
+
+/*---------------------------------------------------------------------------*\
+                                                       
+  FUNCTION....: codec2_decode	     
+  AUTHOR......: David Rowe			      
+  DATE CREATED: 21/8/2010 
+
+  Decodes frames of 51 bits into 160 samples (20ms) of speech.
+
+\*---------------------------------------------------------------------------*/
+
+void codec2_decode(void *codec2_state, short speech[],
+                   const unsigned char * bits)
+{
+    CODEC2 *c2;
+    MODEL   model;
+    int     voiced1, voiced2;
+    int     lsp_indexes[LPC_ORD];
+    int     lpc_correction;
+    int     energy_index;
+    int     Wo_index;
+    float   ak[LPC_ORD+1];
+    int     i;
+    unsigned int nbit = 0;
+    MODEL   model_interp;
+
+    assert(codec2_state != NULL);
+    c2 = (CODEC2*)codec2_state;
+
+    Wo_index = unpack(bits, &nbit, WO_BITS);
+    for(i=0; i<LPC_ORD; i++) {
+	lsp_indexes[i] = unpack(bits, &nbit, lsp_bits(i));
+    }
+    lpc_correction = unpack(bits, &nbit, 1);
+    energy_index = unpack(bits, &nbit, E_BITS);
+    voiced1 = unpack(bits, &nbit, 1);
+    voiced2 = unpack(bits, &nbit, 1);
+    assert(nbit == CODEC2_BITS_PER_FRAME);
+
+    model.Wo = decode_Wo(Wo_index);
+    model.L = PI/model.Wo;
+    decode_amplitudes(&model, 
+		      ak,
+		      lsp_indexes,
+		      lpc_correction, 
+		      energy_index);
+
+    model.voiced = voiced2;
+    model_interp.voiced = voiced1;
+    interpolate(&model_interp, &c2->prev_model, &model);
+
+    synthesise_one_frame(c2,  speech,     &model_interp, ak);
+    synthesise_one_frame(c2, &speech[N],  &model, ak);
+
+    memcpy(&c2->prev_model, &model, sizeof(MODEL));
+}
+
+/*---------------------------------------------------------------------------*\
+                                                       
+  FUNCTION....: synthesise_one_frame()	     
+  AUTHOR......: David Rowe			      
+  DATE CREATED: 23/8/2010 
+
+  Synthesise 80 speech samples (10ms) from model parameters.
+
+\*---------------------------------------------------------------------------*/
+
+void synthesise_one_frame(CODEC2 *c2, short speech[], MODEL *model, float ak[])
+{
+    int     i;
+
+    phase_synth_zero_order(model, ak, &c2->ex_phase);
+    postfilter(model, &c2->bg_est);
+    synthesise(c2->Sn_, model, c2->Pn, 1);
+
+    for(i=0; i<N; i++) {
+	if (c2->Sn_[i] > 32767.0)
+	    speech[i] = 32767;
+	else if (c2->Sn_[i] < -32767.0)
+	    speech[i] = -32767;
+	else
+	    speech[i] = c2->Sn_[i];
+    }
+
+}
+
+/*---------------------------------------------------------------------------*\
+                                                       
+  FUNCTION....: analyse_one_frame()   
+  AUTHOR......: David Rowe			      
+  DATE CREATED: 23/8/2010 
+
+  Extract sinusoidal model parameters from 80 speech samples (10ms of
+  speech).
+ 
+\*---------------------------------------------------------------------------*/
+
+void analyse_one_frame(CODEC2 *c2, MODEL *model, short speech[])
+{
+    COMP    Sw[FFT_ENC];
+    COMP    Sw_[FFT_ENC];
+    float   pitch;
+    int     i;
+
+    /* Read input speech */
+
+    for(i=0; i<M-N; i++)
+      c2->Sn[i] = c2->Sn[i+N];
+    for(i=0; i<N; i++)
+      c2->Sn[i+M-N] = speech[i];
+    dft_speech(Sw, c2->Sn, c2->w);
+
+    /* Estimate pitch */
+
+    nlp(c2->nlp,c2->Sn,N,M,P_MIN,P_MAX,&pitch,Sw,&c2->prev_Wo);
+    c2->prev_Wo = TWO_PI/pitch;
+    model->Wo = TWO_PI/pitch;
+    model->L = PI/model->Wo;
+
+    /* estimate model parameters */
+
+    dft_speech(Sw, c2->Sn, c2->w); 
+    two_stage_pitch_refinement(model, Sw);
+    estimate_amplitudes(model, Sw, c2->W);
+    est_voicing_mbe(model, Sw, c2->W, (FS/TWO_PI)*model->Wo, Sw_);
+}

libs/libcodec2/src/codec2.h

@@ -0,0 +1,43 @@
+/*---------------------------------------------------------------------------*\
+
+  FILE........: codec2.h
+  AUTHOR......: David Rowe
+  DATE CREATED: 21/8/2010
+
+  Codec2 fully quantised encoder and decoder functions.  If you want use 
+  codec2, these are the functions you need to call.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2010 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#ifndef __CODEC2__
+#define  __CODEC2__
+#include "codebook.h"
+
+#define CODEC2_SAMPLES_PER_FRAME 160
+#define CODEC2_BITS_PER_FRAME     51
+
+void *codec2_create();
+void codec2_destroy(void *codec2_state);
+void codec2_encode(void *codec2_state, unsigned char * bits, short speech_in[]);
+void codec2_decode(void *codec2_state, short speech_out[],
+      const unsigned char * bits);
+
+#endif

libs/libcodec2/src/comp.h

@@ -0,0 +1,39 @@
+/*---------------------------------------------------------------------------*\
+                                                                             
+  FILE........: comp.h
+  AUTHOR......: David Rowe                                                          
+  DATE CREATED: 24/08/09
+                                                                             
+  Complex number definition.
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#ifndef __COMP__
+#define __COMP__
+
+/* Complex number */
+
+typedef struct {
+  float real;
+  float imag;
+} COMP;
+
+#endif

libs/libcodec2/src/defines.h

@@ -0,0 +1,84 @@
+/*---------------------------------------------------------------------------*\
+
+  FILE........: defines.h                                                     
+  AUTHOR......: David Rowe                                                          
+  DATE CREATED: 23/4/93                                                       
+                                                                             
+  Defines and structures used throughout the codec.			     
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#ifndef __DEFINES__
+#define __DEFINES__
+
+/*---------------------------------------------------------------------------*\
+                                                                             
+				DEFINES                                       
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+/* General defines */
+
+#define N          80		/* number of samples per frame          */
+#define MAX_AMP    80		/* maximum number of harmonics          */
+#define PI         3.141592654	/* mathematical constant                */
+#define TWO_PI     6.283185307	/* mathematical constant                */
+#define FS         8000		/* sample rate in Hz                    */
+#define MAX_STR    256          /* maximum string size                  */
+
+#define NW         279          /* analysis window size                 */
+#define FFT_ENC    512		/* size of FFT used for encoder         */
+#define FFT_DEC    512	    	/* size of FFT used in decoder          */
+#define TW         40		/* Trapezoidal synthesis window overlap */
+#define V_THRESH   4.0          /* voicing threshold in dB              */
+#define LPC_MAX    20		/* maximum LPC order                    */
+#define LPC_ORD    10		/* phase modelling LPC order            */
+
+/* Pitch estimation defines */
+
+#define M        320		/* pitch analysis frame size            */
+#define P_MIN    20		/* minimum pitch                        */
+#define P_MAX    160		/* maximum pitch                        */
+
+/*---------------------------------------------------------------------------*\
+                                                                             
+				TYPEDEFS                                      
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+/* Complex number */
+
+typedef struct {
+  float real;
+  float imag;
+} COMP;
+
+/* Structure to hold model parameters for one frame */
+
+typedef struct {
+  float Wo;		/* fundamental frequency estimate in radians  */
+  int   L;		/* number of harmonics                        */
+  float A[MAX_AMP];	/* amplitiude of each harmonic                */
+  float phi[MAX_AMP];	/* phase of each harmonic                     */
+  int   voiced;	        /* non-zero if this frame is voiced           */
+} MODEL;
+
+#endif

libs/libcodec2/src/dump.c

@@ -0,0 +1,402 @@
+/*---------------------------------------------------------------------------*\
+                                                                             
+  FILE........: dump.c
+  AUTHOR......: David Rowe                                                          
+  DATE CREATED: 25/8/09                                                       
+                                                                             
+  Routines to dump data to text files for Octave analysis.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#include "defines.h"
+#include "dump.h"
+#include <assert.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+
+static int dumpon = 0;
+
+static FILE *fsn = NULL;
+static FILE *fsw = NULL;
+static FILE *fsw_ = NULL;
+static FILE *fmodel = NULL;
+static FILE *fqmodel = NULL;
+static FILE *fpw = NULL;
+static FILE *flsp = NULL;
+static FILE *fphase = NULL;
+static FILE *fphase_ = NULL;
+static FILE *ffw = NULL;
+static FILE *fe = NULL;
+static FILE *fsq = NULL;
+static FILE *fdec = NULL;
+static FILE *fsnr = NULL;
+static FILE *fak = NULL;
+static FILE *fbg = NULL;
+static FILE *fE = NULL;
+
+static char  prefix[MAX_STR];
+
+void dump_on(char p[]) {
+    dumpon = 1;
+    strcpy(prefix, p);
+}
+
+void dump_off(){
+    if (fsn != NULL)
+	fclose(fsn);
+    if (fsw != NULL)
+	fclose(fsw);
+    if (fsw_ != NULL)
+	fclose(fsw_);
+    if (fmodel != NULL)
+	fclose(fmodel);
+    if (fqmodel != NULL)
+	fclose(fqmodel);
+    if (fpw != NULL)
+	fclose(fpw);
+    if (flsp != NULL)
+	fclose(flsp);
+    if (fphase != NULL)
+	fclose(fphase);
+    if (fphase_ != NULL)
+	fclose(fphase_);
+    if (ffw != NULL)
+	fclose(ffw);
+    if (fe != NULL)
+	fclose(fe);
+    if (fsq != NULL)
+	fclose(fsq);
+    if (fdec != NULL)
+	fclose(fdec);
+    if (fsnr != NULL)
+	fclose(fsnr);
+    if (fak != NULL)
+	fclose(fak);
+    if (fbg != NULL)
+	fclose(fbg);
+    if (fE != NULL)
+	fclose(fE);
+}
+
+void dump_Sn(float Sn[]) {
+    int i;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fsn == NULL) {
+	sprintf(s,"%s_sn.txt", prefix);
+	fsn = fopen(s, "wt");
+	assert(fsn != NULL);
+    }
+
+    /* split across two lines to avoid max line length problems */
+    /* reconstruct in Octave */
+
+    for(i=0; i<M/2; i++)
+	fprintf(fsn,"%f\t",Sn[i]);
+    fprintf(fsn,"\n");    
+    for(i=M/2; i<M; i++)
+	fprintf(fsn,"%f\t",Sn[i]);
+    fprintf(fsn,"\n");    
+}
+
+void dump_Sw(COMP Sw[]) {
+    int i;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fsw == NULL) {
+	sprintf(s,"%s_sw.txt", prefix);
+	fsw = fopen(s, "wt");
+	assert(fsw != NULL);
+    }
+
+    for(i=0; i<FFT_ENC/2; i++)
+	fprintf(fsw,"%f\t",
+		10.0*log10(Sw[i].real*Sw[i].real + Sw[i].imag*Sw[i].imag));
+    fprintf(fsw,"\n");    
+}
+
+void dump_Sw_(COMP Sw_[]) {
+    int i;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fsw_ == NULL) {
+	sprintf(s,"%s_sw_.txt", prefix);
+	fsw_ = fopen(s, "wt");
+	assert(fsw_ != NULL);
+    }
+
+    for(i=0; i<FFT_ENC/2; i++)
+	fprintf(fsw_,"%f\t",
+		10.0*log10(Sw_[i].real*Sw_[i].real + Sw_[i].imag*Sw_[i].imag));
+    fprintf(fsw_,"\n");    
+}
+
+void dump_model(MODEL *model) {
+    int l;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fmodel == NULL) {
+	sprintf(s,"%s_model.txt", prefix);
+	fmodel = fopen(s, "wt");
+	assert(fmodel != NULL);
+    }
+
+    fprintf(fmodel,"%f\t%d\t", model->Wo, model->L);    
+    for(l=1; l<=model->L; l++)
+	fprintf(fmodel,"%f\t",model->A[l]);
+    for(l=model->L+1; l<MAX_AMP; l++)
+	fprintf(fmodel,"0.0\t");
+    fprintf(fmodel,"%d\t",model->voiced);
+    fprintf(fmodel,"\n");    
+}
+
+void dump_quantised_model(MODEL *model) {
+    int l;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fqmodel == NULL) {
+	sprintf(s,"%s_qmodel.txt", prefix);
+	fqmodel = fopen(s, "wt");
+	assert(fqmodel != NULL);
+    }
+
+    fprintf(fqmodel,"%f\t%d\t", model->Wo, model->L);    
+    for(l=1; l<=model->L; l++)
+	fprintf(fqmodel,"%f\t",model->A[l]);
+    for(l=model->L+1; l<MAX_AMP; l++)
+	fprintf(fqmodel,"0.0\t");
+    fprintf(fqmodel,"\n");    
+}
+
+void dump_phase(float phase[], int L) {
+    int l;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fphase == NULL) {
+	sprintf(s,"%s_phase.txt", prefix);
+	fphase = fopen(s, "wt");
+	assert(fphase != NULL);
+    }
+
+    for(l=1; l<=L; l++)
+	fprintf(fphase,"%f\t",phase[l]);
+    for(l=L+1; l<MAX_AMP; l++)
+	fprintf(fphase,"%f\t",0.0);
+    fprintf(fphase,"\n");    
+}
+
+void dump_phase_(float phase_[], int L) {
+    int l;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fphase_ == NULL) {
+	sprintf(s,"%s_phase_.txt", prefix);
+	fphase_ = fopen(s, "wt");
+	assert(fphase_ != NULL);
+    }
+
+    for(l=1; l<=L; l++)
+	fprintf(fphase_,"%f\t",phase_[l]);
+    for(l=L+1; l<MAX_AMP; l++)
+	fprintf(fphase_,"%f\t",0.0);
+    fprintf(fphase_,"\n");    
+}
+
+void dump_snr(float snr) {
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fsnr == NULL) {
+	sprintf(s,"%s_snr.txt", prefix);
+	fsnr = fopen(s, "wt");
+	assert(fsnr != NULL);
+    }
+
+    fprintf(fsnr,"%f\n",snr);
+}
+
+void dump_Pw(COMP Pw[]) {
+    int i;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fpw == NULL) {
+	sprintf(s,"%s_pw.txt", prefix);
+	fpw = fopen(s, "wt");
+	assert(fpw != NULL);
+    }
+
+    for(i=0; i<FFT_DEC/2; i++)
+	fprintf(fpw,"%f\t",Pw[i].real);
+    fprintf(fpw,"\n");    
+}
+
+void dump_lsp(float lsp[]) {
+    int i;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (flsp == NULL) {
+	sprintf(s,"%s_lsp.txt", prefix);
+	flsp = fopen(s, "wt");
+	assert(flsp != NULL);
+    }
+
+    for(i=0; i<10; i++)
+	fprintf(flsp,"%f\t",lsp[i]);
+    fprintf(flsp,"\n");    
+}
+
+void dump_ak(float ak[], int order) {
+    int i;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fak == NULL) {
+	sprintf(s,"%s_ak.txt", prefix);
+	fak = fopen(s, "wt");
+	assert(fak != NULL);
+    }
+
+    for(i=0; i<=order; i++)
+	fprintf(fak,"%f\t",ak[i]);
+    fprintf(fak,"\n");    
+}
+
+void dump_Fw(COMP Fw[]) {
+    int i;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (ffw == NULL) {
+	sprintf(s,"%s_fw.txt", prefix);
+	ffw = fopen(s, "wt");
+	assert(ffw != NULL);
+    }
+
+    for(i=0; i<256; i++)
+	fprintf(ffw,"%f\t",Fw[i].real);
+    fprintf(ffw,"\n");    
+}
+
+void dump_e(float e_hz[]) {
+    int i;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fe == NULL) {
+	sprintf(s,"%s_e.txt", prefix);
+	fe = fopen(s, "wt");
+	assert(fe != NULL);
+    }
+
+    for(i=0; i<500/2; i++)
+	fprintf(fe,"%f\t",e_hz[i]);
+    fprintf(fe,"\n");    
+    for(i=500/2; i<500; i++)
+	fprintf(fe,"%f\t",e_hz[i]);
+    fprintf(fe,"\n");    
+}
+
+void dump_sq(float sq[]) {
+    int i;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fsq == NULL) {
+	sprintf(s,"%s_sq.txt", prefix);
+	fsq = fopen(s, "wt");
+	assert(fsq != NULL);
+    }
+
+    for(i=0; i<M/2; i++)
+	fprintf(fsq,"%f\t",sq[i]);
+    fprintf(fsq,"\n");    
+    for(i=M/2; i<M; i++)
+	fprintf(fsq,"%f\t",sq[i]);
+    fprintf(fsq,"\n");    
+}
+
+void dump_dec(COMP Fw[]) {
+    int i;
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fdec == NULL) {
+	sprintf(s,"%s_dec.txt", prefix);
+	fdec = fopen(s, "wt");
+	assert(fdec != NULL);
+    }
+
+    for(i=0; i<320/5; i++)
+	fprintf(fdec,"%f\t",Fw[i].real);
+    fprintf(fdec,"\n");    
+}
+
+void dump_bg(float e, float bg_est, float percent_uv) {
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fbg == NULL) {
+	sprintf(s,"%s_bg.txt", prefix);
+	fbg = fopen(s, "wt");
+	assert(fbg != NULL);
+    }
+
+    fprintf(fbg,"%f\t%f\t%f\n", e, bg_est, percent_uv);    
+}
+
+void dump_E(float E) {
+    char s[MAX_STR];
+
+    if (!dumpon) return;
+
+    if (fE == NULL) {
+	sprintf(s,"%s_E.txt", prefix);
+	fE = fopen(s, "wt");
+	assert(fE != NULL);
+    }
+
+    fprintf(fE,"%f\n", 10.0*log10(E));
+}

libs/libcodec2/src/dump.h

@@ -0,0 +1,63 @@
+/*---------------------------------------------------------------------------*\
+                                                                             
+  FILE........: dump.h
+  AUTHOR......: David Rowe                                                          
+  DATE CREATED: 25/8/09                                                       
+                                                                             
+  Routines to dump data to text files for Octave analysis.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#ifndef __DUMP__
+#define __DUMP__
+
+void dump_on(char filename_prefix[]);
+void dump_off();
+
+void dump_Sn(float Sn[]);
+void dump_Sw(COMP Sw[]);
+void dump_Sw_(COMP Sw_[]);
+
+/* amplitude modelling */
+
+void dump_model(MODEL *m);
+void dump_quantised_model(MODEL *m);
+void dump_Pw(COMP Pw[]);
+void dump_lsp(float lsp[]);
+void dump_ak(float ak[], int order);
+void dump_E(float E);
+
+/* phase modelling */
+
+void dump_snr(float snr);
+void dump_phase(float phase[], int L);
+void dump_phase_(float phase[], int L);
+
+/* NLP states */
+
+void dump_sq(float sq[]);
+void dump_dec(COMP Fw[]);
+void dump_Fw(COMP Fw[]);
+void dump_e(float e_hz[]);
+
+/* post filter */
+
+void dump_bg(float e, float bg_est, float percent_uv);
+
+#endif

libs/libcodec2/src/four1.c

@@ -0,0 +1,64 @@
+/*---------------------------------------------------------------------------*\
+  
+  FILE........: four1.c
+  AUTHOR......: David Rowe
+  DATE CREATED: 20/2/95
+
+  Numerical Recipies in C FFT function.  I have a nasty licence so please
+  replace me.
+
+\*---------------------------------------------------------------------------*/
+
+#include <math.h>
+
+#define SWAP(a,b) tempr=(a);(a)=(b);(b)=tempr
+
+void four1(data,nn,isign)
+float data[];
+int nn,isign;
+{
+	int n,mmax,m,j,istep,i;
+	double wtemp,wr,wpr,wpi,wi,theta;
+	float tempr,tempi;
+
+	n=nn << 1;
+	j=1;
+	for (i=1;i<n;i+=2) {
+		if (j > i) {
+			SWAP(data[j],data[i]);
+			SWAP(data[j+1],data[i+1]);
+		}
+		m=n >> 1;
+		while (m >= 2 && j > m) {
+			j -= m;
+			m >>= 1;
+		}
+		j += m;
+	}
+	mmax=2;
+	while (n > mmax) {
+		istep=2*mmax;
+		theta=6.28318530717959/(isign*mmax);
+		wtemp=sin(0.5*theta);
+		wpr = -2.0*wtemp*wtemp;
+		wpi=sin(theta);
+		wr=1.0;
+		wi=0.0;
+		for (m=1;m<mmax;m+=2) {
+			for (i=m;i<=n;i+=istep) {
+				j=i+mmax;
+				tempr=wr*data[j]-wi*data[j+1];
+				tempi=wr*data[j+1]+wi*data[j];
+				data[j]=data[i]-tempr;
+				data[j+1]=data[i+1]-tempi;
+				data[i] += tempr;
+				data[i+1] += tempi;
+			}
+			wr=(wtemp=wr)*wpr-wi*wpi+wr;
+			wi=wi*wpr+wtemp*wpi+wi;
+		}
+		mmax=istep;
+	}
+}
+
+#undef SWAP

libs/libcodec2/src/four1.h

@@ -0,0 +1,18 @@
+/*---------------------------------------------------------------------------*\
+  
+  FILE........: four1.h
+  AUTHOR......: David Rowe
+  DATE CREATED: 20/2/95
+
+  Numerical Recipies in C FFT function.  I have a nasty licence so please
+  replace me.
+
+\*---------------------------------------------------------------------------*/
+
+#ifndef __FOUR1__
+#define __FOUR1__
+
+void four1(float x[], int n, int isign);
+
+#endif	/* __FOUR1__ */
+

libs/libcodec2/src/fq20.sh

@@ -0,0 +1,8 @@
+#!/bin/sh
+# fq20.shsh
+# David Rowe 27 July 2010
+# 
+# Decode a file with fully quantised codec at 20ms frame rate
+
+../src/sinedec ../raw/$1.raw $1.mdl -o $1_phase0_lsp_20_EWo2.raw --phase 0 --lpc 10 --lsp --postfilter --dec
+

libs/libcodec2/src/globals.c

@@ -0,0 +1,50 @@
+/*---------------------------------------------------------------------------*\
+                                        
+  FILE........: globals.c
+  AUTHOR......: David Rowe                                                          
+  DATE CREATED: 11/5/94                          
+                                                                             
+  Globals for sinusoidal speech coder.		      
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#include "sine.h"	/* global defines for coder */
+
+/* Globals used in encoder and decoder */
+
+int frames;		/* number of frames processed so far */
+float Sn[M];	        /* float input speech samples */
+MODEL model;		/* model parameters for the current frame */
+int Nw;			/* number of samples in analysis window */
+float sig;		/* energy of current frame */
+
+/* Globals used in encoder */
+
+float w[M];	        /* time domain hamming window */
+COMP W[FFT_ENC];	/* DFT of w[] */
+COMP Sw[FFT_ENC];	/* DFT of current frame */
+
+/* Globals used in decoder */
+
+COMP Sw_[FFT_ENC];	/* DFT of all voiced synthesised signal */
+float Sn_[AW_DEC];	/* synthesised speech */
+float Pn[AW_DEC];	/* time domain Parzen (trapezoidal) window */
+

libs/libcodec2/src/globals.h

@@ -0,0 +1,48 @@
+/*---------------------------------------------------------------------------*\
+                                                                             
+  FILE........: globals.h
+  AUTHOR......: David Rowe
+  DATE CREATED: 1/11/94                                                       
+                                                                             
+  Globals for sinusoidal speech coder.					      
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+/* Globals used in encoder and decoder */
+
+extern int frames;	/* number of frames processed so far */
+extern float Sn[];	/* float input speech samples */
+extern MODEL model;	/* model parameters for the current frame */
+extern int Nw;		/* number of samples in analysis window */
+extern float sig;	/* energy of current frame */
+
+/* Globals used in encoder */
+
+extern float w[];	/* time domain hamming window */
+extern COMP W[];	/* frequency domain hamming window */
+extern COMP Sw[];	/* DFT of current frame */
+extern COMP Sw_[];	/* DFT of all voiced synthesised signal */
+
+/* Globals used in decoder */
+
+extern float Sn_[];	/* output synthesised speech samples */
+extern float Pn[];	/* time domain Parzen (trapezoidal) window */
+

libs/libcodec2/src/interp.c

@@ -0,0 +1,122 @@
+/*---------------------------------------------------------------------------*\
+
+  FILE........: interp.c
+  AUTHOR......: David Rowe
+  DATE CREATED: 9/10/09
+
+  Interpolation of 20ms frames to 10ms frames.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#include <assert.h>
+#include <math.h>
+#include <string.h>
+
+#include "defines.h"
+#include "interp.h"
+
+float sample_log_amp(MODEL *model, float w);
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: interp()	     
+  AUTHOR......: David Rowe			      
+  DATE CREATED: 22/8/10 
+        
+  Given two frames decribed by model parameters 20ms apart, determines
+  the model parameters of the 10ms frame between them.  Assumes
+  voicing is available for middle (interpolated) frame.  Outputs are
+  amplitudes and Wo for the interpolated frame.
+
+  This version can interpolate the amplitudes between two frames of
+  different Wo and L.
+  
+\*---------------------------------------------------------------------------*/
+
+void interpolate(
+  MODEL *interp,    /* interpolated model params                     */
+  MODEL *prev,      /* previous frames model params                  */
+  MODEL *next       /* next frames model params                      */
+)
+{
+    int   l;
+    float w,log_amp;
+
+    /* Wo depends on voicing of this and adjacent frames */
+
+    if (interp->voiced) {
+	if (prev->voiced && next->voiced)
+	    interp->Wo = (prev->Wo + next->Wo)/2.0;
+	if (!prev->voiced && next->voiced)
+	    interp->Wo = next->Wo;
+	if (prev->voiced && !next->voiced)
+	    interp->Wo = prev->Wo;
+    }
+    else {
+	interp->Wo = TWO_PI/P_MAX;
+    }
+    interp->L = PI/interp->Wo;
+
+    /* Interpolate amplitudes using linear interpolation in log domain */
+
+    for(l=1; l<=interp->L; l++) {
+	w = l*interp->Wo;
+	log_amp = (sample_log_amp(prev, w) + sample_log_amp(next, w))/2.0;
+	interp->A[l] = pow(10.0, log_amp);
+    }
+}
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: sample_log_amp()
+  AUTHOR......: David Rowe			      
+  DATE CREATED: 22/8/10 
+        
+  Samples the amplitude envelope at an arbitrary frequency w.  Uses
+  linear interpolation in the log domain to sample between harmonic
+  amplitudes.
+  
+\*---------------------------------------------------------------------------*/
+
+float sample_log_amp(MODEL *model, float w)
+{
+    int   m;
+    float f, log_amp;
+
+    assert(w > 0.0); assert (w <= PI);
+
+    m = floor(w/model->Wo + 0.5);
+    f = (w - m*model->Wo)/w;
+    assert(f <= 1.0);
+
+    if (m < 1) {
+	log_amp = f*log10(model->A[1]);
+    }
+    else if ((m+1) > model->L) {
+	log_amp = (1.0-f)*log10(model->A[model->L]);
+    }
+    else {
+	log_amp = (1.0-f)*log10(model->A[m]) + f*log10(model->A[m+1]);
+    }
+
+    return log_amp;
+}
+

libs/libcodec2/src/interp.h

@@ -0,0 +1,34 @@
+/*---------------------------------------------------------------------------*\
+
+  FILE........: interp.h
+  AUTHOR......: David Rowe
+  DATE CREATED: 9/10/09
+
+  Interpolation of 20ms frames to 10ms frames.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#ifndef __INTERP__
+#define __INTERP__
+
+void interpolate(MODEL *interp, MODEL *prev, MODEL *next);
+
+#endif

libs/libcodec2/src/listen.sh

@@ -0,0 +1,9 @@
+#!/bin/sh
+# listensim.sh
+# David Rowe 10 Sep 2009
+#
+# Listen to files processed with sim.sh
+
+../script/menu.sh ../raw/$1.raw $1_uq.raw $1_phase0.raw $1_lpc10.raw $1_lsp.raw $1_phase0_lpc10.raw $1_phase0_lsp.raw $1_phase0_lsp.raw $2 $3
+
+

libs/libcodec2/src/listen1.sh

@@ -0,0 +1,15 @@
+#!/bin/sh
+# listen1.sh
+# David Rowe 10 Sep 2009
+#
+# Run menu with common sample file options, headphone version
+
+#../script/menu.sh ../raw/$1.raw $1_uq.raw $1_phase0.raw $1_lpc10.raw $1_lsp.raw $1_phase0_lpc10.raw $1_phase0_lsp.raw ../raw/$1_g729a.raw $2 $3 -d /dev/dsp1
+
+# compare to other codecs
+
+#../script/menu.sh ../raw/$1.raw $1_phase0_lsp.raw $1_phase0_lsp_20.raw ../raw/$1_g729a.raw ../raw/$1_gsm13k.raw ../raw/$1_speex_8k.raw $2 $3 -d /dev/dsp1
+
+../script/menu.sh ../raw/$1.raw $1_uq.raw $1_phase0.raw $1_test.raw ../raw/$1_g729a.raw $2 $3 -d /dev/dsp1
+
+

libs/libcodec2/src/listensim.sh

@@ -0,0 +1,9 @@
+#!/bin/sh
+# listensim.sh
+# David Rowe 10 Sep 2009
+#
+# Listen to files processed with sim.sh
+
+../script/menu.sh ../raw/$1.raw $1_uq.raw $1_phase0.raw $1_lpc10.raw $1_lsp.raw $1_phase0_lpc10.raw $1_phase0_lsp.raw $1_phase0_lsp_dec.raw $2 $3
+
+

libs/libcodec2/src/lpc.c

@@ -0,0 +1,253 @@
+/*---------------------------------------------------------------------------*\
+                                                                           
+  FILE........: lpc.c                                                              
+  AUTHOR......: David Rowe                                                      
+  DATE CREATED: 30/9/90                                                   
+                                                                          
+  Linear Prediction functions written in C.                                
+                                                                          
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#define LPC_MAX_N 512		/* maximum no. of samples in frame */
+#define PI 3.141592654		/* mathematical constant */
+
+#include <assert.h>
+#include <math.h>
+#include "defines.h"
+#include "lpc.h"
+
+/*---------------------------------------------------------------------------*\
+                                                                         
+  hanning_window()                                                        
+                                                                         
+  Hanning windows a frame of speech samples.                              
+                                                                           
+\*---------------------------------------------------------------------------*/
+
+void hanning_window(
+  float Sn[],	/* input frame of speech samples */
+  float Wn[],	/* output frame of windowed samples */
+  int Nsam	/* number of samples */
+)
+{
+  int i;	/* loop variable */
+
+  for(i=0; i<Nsam; i++)
+    Wn[i] = Sn[i]*(0.5 - 0.5*cos(2*PI*(float)i/(Nsam-1)));
+}
+
+/*---------------------------------------------------------------------------*\
+                                                                           
+  autocorrelate()                                                          
+                                                                          
+  Finds the first P autocorrelation values of an array of windowed speech 
+  samples Sn[].                                                            
+                                                                          
+\*---------------------------------------------------------------------------*/
+
+void autocorrelate(
+  float Sn[],	/* frame of Nsam windowed speech samples */
+  float Rn[],	/* array of P+1 autocorrelation coefficients */
+  int Nsam,	/* number of windowed samples to use */
+  int order	/* order of LPC analysis */
+)
+{
+  int i,j;	/* loop variables */
+
+  for(j=0; j<order+1; j++) {
+    Rn[j] = 0.0;
+    for(i=0; i<Nsam-j; i++)
+      Rn[j] += Sn[i]*Sn[i+j];
+  }
+}
+
+/*---------------------------------------------------------------------------*\
+                                                                            
+  levinson_durbin()                                                        
+                                                                           
+  Given P+1 autocorrelation coefficients, finds P Linear Prediction Coeff. 
+  (LPCs) where P is the order of the LPC all-pole model. The Levinson-Durbin
+  algorithm is used, and is described in:                                   
+                                                                           
+    J. Makhoul                                                               
+    "Linear prediction, a tutorial review"                                   
+    Proceedings of the IEEE                                                
+    Vol-63, No. 4, April 1975                                               
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+void levinson_durbin(
+  float R[],		/* order+1 autocorrelation coeff */
+  float lpcs[],		/* order+1 LPC's */
+  int order		/* order of the LPC analysis */
+)
+{
+  float E[LPC_MAX+1];
+  float k[LPC_MAX+1];
+  float a[LPC_MAX+1][LPC_MAX+1];
+  float sum;
+  int i,j;				/* loop variables */
+
+  E[0] = R[0];				/* Equation 38a, Makhoul */
+
+  for(i=1; i<=order; i++) {
+    sum = 0.0;
+    for(j=1; j<=i-1; j++)
+      sum += a[i-1][j]*R[i-j];
+    k[i] = -1.0*(R[i] + sum)/E[i-1];	/* Equation 38b, Makhoul */
+    if (fabs(k[i]) > 1.0)
+      k[i] = 0.0;
+
+    a[i][i] = k[i];
+
+    for(j=1; j<=i-1; j++)
+      a[i][j] = a[i-1][j] + k[i]*a[i-1][i-j];	/* Equation 38c, Makhoul */
+
+    E[i] = (1-k[i]*k[i])*E[i-1];		/* Equation 38d, Makhoul */
+  }
+
+  for(i=1; i<=order; i++)
+    lpcs[i] = a[order][i];
+  lpcs[0] = 1.0;  
+}
+
+/*---------------------------------------------------------------------------*\
+                                                                             
+  inverse_filter()                                                          
+                                                                           
+  Inverse Filter, A(z).  Produces an array of residual samples from an array
+  of input samples and linear prediction coefficients.                      
+                                                                             
+  The filter memory is stored in the first order samples of the input array.
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+void inverse_filter(
+  float Sn[],	/* Nsam input samples */
+  float a[],	/* LPCs for this frame of samples */
+  int Nsam,	/* number of samples */
+  float res[],	/* Nsam residual samples */
+  int order	/* order of LPC */
+)
+{
+  int i,j;	/* loop variables */
+
+  for(i=0; i<Nsam; i++) {
+    res[i] = 0.0;
+    for(j=0; j<=order; j++)
+      res[i] += Sn[i-j]*a[j];
+  }    
+}
+
+/*---------------------------------------------------------------------------*\
+                                                                            
+ synthesis_filter()                                                        
+                                                                           
+ C version of the Speech Synthesis Filter, 1/A(z).  Given an array of   
+ residual or excitation samples, and the the LP filter coefficients, this
+ function will produce an array of speech samples.  This filter structure is
+ IIR.                                                                       
+                                                                           
+ The synthesis filter has memory as well, this is treated in the same way 
+ as the memory for the inverse filter (see inverse_filter() notes above). 
+ The difference is that the memory for the synthesis filter is stored in  
+ the output array, wheras the memory of the inverse filter is stored in the
+ input array.                                                              
+                                                                           
+ Note: the calling function must update the filter memory.                
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+void synthesis_filter(
+  float res[],	/* Nsam input residual (excitation) samples */
+  float a[],	/* LPCs for this frame of speech samples */
+  int Nsam,	/* number of speech samples */
+  int order,	/* LPC order */
+  float Sn_[]	/* Nsam output synthesised speech samples */
+)
+{
+  int i,j;	/* loop variables */
+
+  /* Filter Nsam samples */
+
+  for(i=0; i<Nsam; i++) {
+    Sn_[i] = res[i]*a[0];
+    for(j=1; j<=order; j++)
+      Sn_[i] -= Sn_[i-j]*a[j];
+  }
+}
+
+/*---------------------------------------------------------------------------*\
+                                                                            
+  find_aks()                                                                 
+                                                                            
+  This function takes a frame of samples, and determines the linear           
+  prediction coefficients for that frame of samples.                         
+                                                                            
+\*---------------------------------------------------------------------------*/
+
+void find_aks(
+  float Sn[],	/* Nsam samples with order sample memory */
+  float a[],	/* order+1 LPCs with first coeff 1.0 */
+  int Nsam,	/* number of input speech samples */
+  int order,	/* order of the LPC analysis */
+  float *E	/* residual energy */
+)
+{
+  float Wn[LPC_MAX_N];	/* windowed frame of Nsam speech samples */
+  float R[LPC_MAX+1];	/* order+1 autocorrelation values of Sn[] */
+  int i;
+
+  assert(order < LPC_MAX);
+  assert(Nsam < LPC_MAX_N);
+
+  hanning_window(Sn,Wn,Nsam);
+  autocorrelate(Wn,R,Nsam,order);
+  levinson_durbin(R,a,order);
+
+  *E = 0.0;
+  for(i=0; i<=order; i++)
+    *E += a[i]*R[i];
+  if (*E < 0.0)
+    *E = 1E-12;
+}
+
+/*---------------------------------------------------------------------------*\
+                                                                            
+  weight()                                                                  
+                                                                          
+  Weights a vector of LPCs.						   
+                                                                          
+\*---------------------------------------------------------------------------*/
+
+void weight(
+  float ak[],	/* vector of order+1 LPCs */
+  float gamma,	/* weighting factor */
+  int order,	/* num LPCs (excluding leading 1.0) */
+  float akw[]	/* weighted vector of order+1 LPCs */
+)
+{
+  int i;
+  
+  for(i=1; i<=order; i++)
+    akw[i] = ak[i]*pow(gamma,(float)i);
+}
+    

libs/libcodec2/src/lpc.h

@@ -0,0 +1,42 @@
+/*---------------------------------------------------------------------------*\
+                                                                           
+  FILE........: lpc.h                                                            
+  AUTHOR......: David Rowe                                                      
+  DATE CREATED: 24/8/09                                                   
+                                                                          
+  Linear Prediction functions written in C.                                
+                                                                          
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#ifndef __LPC__
+#define __LPC__
+
+#define LPC_MAX_ORDER 20
+
+void hanning_window(float Sn[],	float Wn[], int Nsam);
+void autocorrelate(float Sn[], float Rn[], int Nsam, int order);
+void levinson_durbin(float R[],	float lpcs[], int order);
+void inverse_filter(float Sn[], float a[], int Nsam, float res[], int order);
+void synthesis_filter(float res[], float a[], int Nsam,	int order, float Sn_[]);
+void find_aks(float Sn[], float a[], int Nsam, int order, float *E);
+void weight(float ak[],	float gamma, int order,	float akw[]);
+
+#endif

libs/libcodec2/src/lsp.c

@@ -0,0 +1,323 @@
+/*---------------------------------------------------------------------------*\
+
+  FILE........: lsp.c
+  AUTHOR......: David Rowe
+  DATE CREATED: 24/2/93
+
+
+  This file contains functions for LPC to LSP conversion and LSP to
+  LPC conversion. Note that the LSP coefficients are not in radians
+  format but in the x domain of the unit circle.
+
+\*---------------------------------------------------------------------------*/
+
+#include "defines.h"
+#include "lsp.h"
+#include <math.h>
+#include <stdio.h>
+#include <stdlib.h>
+
+/*---------------------------------------------------------------------------*\
+
+  Introduction to Line Spectrum Pairs (LSPs)
+  ------------------------------------------
+
+  LSPs are used to encode the LPC filter coefficients {ak} for
+  transmission over the channel.  LSPs have several properties (like
+  less sensitivity to quantisation noise) that make them superior to
+  direct quantisation of {ak}.
+
+  A(z) is a polynomial of order lpcrdr with {ak} as the coefficients.
+
+  A(z) is transformed to P(z) and Q(z) (using a substitution and some
+  algebra), to obtain something like:
+
+    A(z) = 0.5[P(z)(z+z^-1) + Q(z)(z-z^-1)]  (1)
+
+  As you can imagine A(z) has complex zeros all over the z-plane. P(z)
+  and Q(z) have the very neat property of only having zeros _on_ the
+  unit circle.  So to find them we take a test point z=exp(jw) and
+  evaluate P (exp(jw)) and Q(exp(jw)) using a grid of points between 0
+  and pi.
+
+  The zeros (roots) of P(z) also happen to alternate, which is why we
+  swap coefficients as we find roots.  So the process of finding the
+  LSP frequencies is basically finding the roots of 5th order
+  polynomials.
+
+  The root so P(z) and Q(z) occur in symmetrical pairs at +/-w, hence
+  the name Line Spectrum Pairs (LSPs).
+
+  To convert back to ak we just evaluate (1), "clocking" an impulse
+  thru it lpcrdr times gives us the impulse response of A(z) which is
+  {ak}.
+
+\*---------------------------------------------------------------------------*/
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: cheb_poly_eva()
+  AUTHOR......: David Rowe
+  DATE CREATED: 24/2/93
+
+  This function evalutes a series of chebyshev polynomials
+
+  FIXME: performing memory allocation at run time is very inefficient,
+  replace with stack variables of MAX_P size.
+
+\*---------------------------------------------------------------------------*/
+
+
+float cheb_poly_eva(float *coef,float x,int m)
+/*  float coef[]  	coefficients of the polynomial to be evaluated 	*/
+/*  float x   		the point where polynomial is to be evaluated 	*/
+/*  int m 		order of the polynomial 			*/
+{
+    int i;
+    float *T,*t,*u,*v,sum;
+
+    /* Allocate memory for chebyshev series formulation */
+
+    if((T = (float *)malloc((m/2+1)*sizeof(float))) == NULL){
+	fprintf(stderr, "not enough memory to allocate buffer\n");
+	exit(1);
+    }
+
+    /* Initialise pointers */
+
+    t = T;                          	/* T[i-2] 			*/
+    *t++ = 1.0;
+    u = t--;                        	/* T[i-1] 			*/
+    *u++ = x;
+    v = u--;                        	/* T[i] 			*/
+
+    /* Evaluate chebyshev series formulation using iterative approach 	*/
+
+    for(i=2;i<=m/2;i++)
+	*v++ = (2*x)*(*u++) - *t++;  	/* T[i] = 2*x*T[i-1] - T[i-2]	*/
+
+    sum=0.0;                        	/* initialise sum to zero 	*/
+    t = T;                          	/* reset pointer 		*/
+
+    /* Evaluate polynomial and return value also free memory space */
+
+    for(i=0;i<=m/2;i++)
+	sum+=coef[(m/2)-i]**t++;
+
+    free(T);
+    return sum;
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: lpc_to_lsp()
+  AUTHOR......: David Rowe
+  DATE CREATED: 24/2/93
+
+  This function converts LPC coefficients to LSP coefficients.
+
+\*---------------------------------------------------------------------------*/
+
+int lpc_to_lsp (float *a, int lpcrdr, float *freq, int nb, float delta)
+/*  float *a 		     	lpc coefficients			*/
+/*  int lpcrdr			order of LPC coefficients (10) 		*/
+/*  float *freq 	      	LSP frequencies in radians      	*/
+/*  int nb			number of sub-intervals (4) 		*/
+/*  float delta			grid spacing interval (0.02) 		*/
+{
+    float psuml,psumr,psumm,temp_xr,xl,xr,xm;
+    float temp_psumr;
+    int i,j,m,flag,k;
+    float *Q;                 	/* ptrs for memory allocation 		*/
+    float *P;
+    float *px;                	/* ptrs of respective P'(z) & Q'(z)	*/
+    float *qx;
+    float *p;
+    float *q;
+    float *pt;                	/* ptr used for cheb_poly_eval()
+				   whether P' or Q' 			*/
+    int roots=0;              	/* number of roots found 	        */
+    flag = 1;                	
+    m = lpcrdr/2;            	/* order of P'(z) & Q'(z) polynimials 	*/
+
+    /* Allocate memory space for polynomials */
+
+    Q = (float *) malloc((m+1)*sizeof(float));
+    P = (float *) malloc((m+1)*sizeof(float));
+    if( (P == NULL) || (Q == NULL) ) {
+	fprintf(stderr,"not enough memory to allocate buffer\n");
+	exit(1);
+    }
+
+    /* determine P'(z)'s and Q'(z)'s coefficients where
+      P'(z) = P(z)/(1 + z^(-1)) and Q'(z) = Q(z)/(1-z^(-1)) */
+
+    px = P;                      /* initilaise ptrs */
+    qx = Q;
+    p = px;
+    q = qx;
+    *px++ = 1.0;
+    *qx++ = 1.0;
+    for(i=1;i<=m;i++){
+	*px++ = a[i]+a[lpcrdr+1-i]-*p++;
+	*qx++ = a[i]-a[lpcrdr+1-i]+*q++;
+    }
+    px = P;
+    qx = Q;
+    for(i=0;i<m;i++){
+	*px = 2**px;
+	*qx = 2**qx;
+	 px++;
+	 qx++;
+    }
+    px = P;             	/* re-initialise ptrs 			*/
+    qx = Q;
+
+    /* Search for a zero in P'(z) polynomial first and then alternate to Q'(z).
+    Keep alternating between the two polynomials as each zero is found 	*/
+
+    xr = 0;             	/* initialise xr to zero 		*/
+    xl = 1.0;               	/* start at point xl = 1 		*/
+
+
+    for(j=0;j<lpcrdr;j++){
+	if(j%2)            	/* determines whether P' or Q' is eval. */
+	    pt = qx;
+	else
+	    pt = px;
+
+	psuml = cheb_poly_eva(pt,xl,lpcrdr);	/* evals poly. at xl 	*/
+	flag = 1;
+	while(flag && (xr >= -1.0)){
+	    xr = xl - delta ;                  	/* interval spacing 	*/
+	    psumr = cheb_poly_eva(pt,xr,lpcrdr);/* poly(xl-delta_x) 	*/
+	    temp_psumr = psumr;
+	    temp_xr = xr;
+
+        /* if no sign change increment xr and re-evaluate
+           poly(xr). Repeat til sign change.  if a sign change has
+           occurred the interval is bisected and then checked again
+           for a sign change which determines in which interval the
+           zero lies in.  If there is no sign change between poly(xm)
+           and poly(xl) set interval between xm and xr else set
+           interval between xl and xr and repeat till root is located
+           within the specified limits  */
+
+	    if((psumr*psuml)<0.0){
+		roots++;
+
+		psumm=psuml;
+		for(k=0;k<=nb;k++){
+		    xm = (xl+xr)/2;        	/* bisect the interval 	*/
+		    psumm=cheb_poly_eva(pt,xm,lpcrdr);
+		    if(psumm*psuml>0.){
+			psuml=psumm;
+			xl=xm;
+		    }
+		    else{
+			psumr=psumm;
+			xr=xm;
+		    }
+		}
+
+	       /* once zero is found, reset initial interval to xr 	*/
+	       freq[j] = (xm);
+	       xl = xm;
+	       flag = 0;       		/* reset flag for next search 	*/
+	    }
+	    else{
+		psuml=temp_psumr;
+		xl=temp_xr;
+	    }
+	}
+    }
+    free(P);                  		/* free memory space 		*/
+    free(Q);
+
+    /* convert from x domain to radians */
+
+    for(i=0; i<lpcrdr; i++) {
+	freq[i] = acos(freq[i]);
+    }
+
+    return(roots);
+}
+
+/*---------------------------------------------------------------------------*\
+
+  FUNCTION....: lsp_to_lpc()
+  AUTHOR......: David Rowe
+  DATE CREATED: 24/2/93
+
+  This function converts LSP coefficients to LPC coefficients.  In the
+  Speex code we worked out a wayto simplify this significantly.
+
+\*---------------------------------------------------------------------------*/
+
+void lsp_to_lpc(float *freq, float *ak, int lpcrdr)
+/*  float *freq         array of LSP frequencies in radians     	*/
+/*  float *ak 		array of LPC coefficients 			*/
+/*  int lpcrdr  	order of LPC coefficients 			*/
+
+
+{
+    int i,j;
+    float xout1,xout2,xin1,xin2;
+    float *Wp;
+    float *pw,*n1,*n2,*n3,*n4;
+    int m = lpcrdr/2;
+
+    /* convert from radians to the x=cos(w) domain */
+
+    for(i=0; i<lpcrdr; i++)
+	freq[i] = cos(freq[i]);
+
+    if((Wp = (float *) malloc((4*m+2)*sizeof(float))) == NULL){
+	printf("not enough memory to allocate buffer\n");
+	exit(1);
+    }
+    pw = Wp;
+
+    /* initialise contents of array */
+
+    for(i=0;i<=4*m+1;i++){       	/* set contents of buffer to 0 */
+	*pw++ = 0.0;
+    }
+
+    /* Set pointers up */
+
+    pw = Wp;
+    xin1 = 1.0;
+    xin2 = 1.0;
+
+    /* reconstruct P(z) and Q(z) by cascading second order polynomials
+      in form 1 - 2xz(-1) +z(-2), where x is the LSP coefficient */
+
+    for(j=0;j<=lpcrdr;j++){
+	for(i=0;i<m;i++){
+	    n1 = pw+(i*4);
+	    n2 = n1 + 1;
+	    n3 = n2 + 1;
+	    n4 = n3 + 1;
+	    xout1 = xin1 - 2*(freq[2*i]) * *n1 + *n2;
+	    xout2 = xin2 - 2*(freq[2*i+1]) * *n3 + *n4;
+	    *n2 = *n1;
+	    *n4 = *n3;
+	    *n1 = xin1;
+	    *n3 = xin2;
+	    xin1 = xout1;
+	    xin2 = xout2;
+	}
+	xout1 = xin1 + *(n4+1);
+	xout2 = xin2 - *(n4+2);
+	ak[j] = (xout1 + xout2)*0.5;
+	*(n4+1) = xin1;
+	*(n4+2) = xin2;
+
+	xin1 = 0.0;
+	xin2 = 0.0;
+    }
+    free(Wp);
+}
+

libs/libcodec2/src/lsp.h

@@ -0,0 +1,20 @@
+/*---------------------------------------------------------------------------*\
+
+  FILE........: lsp.c
+  AUTHOR......: David Rowe
+  DATE CREATED: 24/2/93
+
+
+  This file contains functions for LPC to LSP conversion and LSP to
+  LPC conversion. Note that the LSP coefficients are not in radians
+  format but in the x domain of the unit circle.
+
+\*---------------------------------------------------------------------------*/
+
+#ifndef __LSP__
+#define __LSP__
+
+int lpc_to_lsp (float *a, int lpcrdr, float *freq, int nb, float delta);
+void lsp_to_lpc(float *freq, float *ak, int lpcrdr);
+
+#endif

libs/libcodec2/src/nlp.c

@@ -0,0 +1,361 @@
+/*---------------------------------------------------------------------------*\
+                                                 
+  FILE........: nlp.c                                                   
+  AUTHOR......: David Rowe                                      
+  DATE CREATED: 23/3/93                                    
+                                                         
+  Non Linear Pitch (NLP) estimation functions.			  
+                                                               
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#include "defines.h"
+#include "nlp.h"
+#include "dump.h"
+#include "four1.h"
+
+#include <assert.h>
+#include <math.h>
+#include <stdlib.h>
+
+/*---------------------------------------------------------------------------*\
+                                                                             
+ 				DEFINES                                       
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+#define PMAX_M      600		/* maximum NLP analysis window size     */
+#define COEFF       0.95	/* notch filter parameter               */
+#define PE_FFT_SIZE 512		/* DFT size for pitch estimation        */
+#define DEC         5		/* decimation factor                    */
+#define SAMPLE_RATE 8000
+#define PI          3.141592654	/* mathematical constant                */
+#define T           0.1         /* threshold for local minima candidate */
+#define F0_MAX      500
+#define CNLP        0.3	        /* post processor constant              */
+#define NLP_NTAP 48	        /* Decimation LPF order */
+
+/*---------------------------------------------------------------------------*\
+                                                                            
+ 				GLOBALS                                       
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+/* 48 tap 600Hz low pass FIR filter coefficients */
+
+float nlp_fir[] = {
+  -1.0818124e-03,
+  -1.1008344e-03,
+  -9.2768838e-04,
+  -4.2289438e-04,
+   5.5034190e-04,
+   2.0029849e-03,
+   3.7058509e-03,
+   5.1449415e-03,
+   5.5924666e-03,
+   4.3036754e-03,
+   8.0284511e-04,
+  -4.8204610e-03,
+  -1.1705810e-02,
+  -1.8199275e-02,
+  -2.2065282e-02,
+  -2.0920610e-02,
+  -1.2808831e-02,
+   3.2204775e-03,
+   2.6683811e-02,
+   5.5520624e-02,
+   8.6305944e-02,
+   1.1480192e-01,
+   1.3674206e-01,
+   1.4867556e-01,
+   1.4867556e-01,
+   1.3674206e-01,
+   1.1480192e-01,
+   8.6305944e-02,
+   5.5520624e-02,
+   2.6683811e-02,
+   3.2204775e-03,
+  -1.2808831e-02,
+  -2.0920610e-02,
+  -2.2065282e-02,
+  -1.8199275e-02,
+  -1.1705810e-02,
+  -4.8204610e-03,
+   8.0284511e-04,
+   4.3036754e-03,
+   5.5924666e-03,
+   5.1449415e-03,
+   3.7058509e-03,
+   2.0029849e-03,
+   5.5034190e-04,
+  -4.2289438e-04,
+  -9.2768838e-04,
+  -1.1008344e-03,
+  -1.0818124e-03
+};
+
+typedef struct {
+    float sq[PMAX_M];	     /* squared speech samples */
+    float mem_x,mem_y;       /* memory for notch filter */
+    float mem_fir[NLP_NTAP]; /* decimation FIR filter memory */
+} NLP;
+
+float post_process_mbe(COMP Fw[], int pmin, int pmax, float gmax);
+float post_process_sub_multiples(COMP Fw[], 
+				 int pmin, int pmax, float gmax, int gmax_bin,
+				 float *prev_Wo);
+
+/*---------------------------------------------------------------------------*\
+                                                                             
+  nlp_create()                                                                  
+                                                                             
+  Initialisation function for NLP pitch estimator.
+
+\*---------------------------------------------------------------------------*/
+
+void *nlp_create()
+{
+    NLP *nlp;
+    int  i;
+
+    nlp = (NLP*)malloc(sizeof(NLP));
+    if (nlp == NULL)
+	return NULL;
+
+    for(i=0; i<PMAX_M; i++)
+	nlp->sq[i] = 0.0;
+    nlp->mem_x = 0.0;
+    nlp->mem_y = 0.0;
+    for(i=0; i<NLP_NTAP; i++)
+	nlp->mem_fir[i] = 0.0;
+
+    return (void*)nlp;
+}
+
+/*---------------------------------------------------------------------------*\
+                                                                             
+  nlp_destory()
+                                                                             
+  Initialisation function for NLP pitch estimator.
+
+\*---------------------------------------------------------------------------*/
+
+void nlp_destroy(void *nlp_state)
+{
+    assert(nlp_state != NULL);
+    free(nlp_state);
+}
+
+/*---------------------------------------------------------------------------*\
+                                                                             
+  nlp()                                                                  
+                                                                             
+  Determines the pitch in samples using the Non Linear Pitch (NLP)
+  algorithm [1]. Returns the fundamental in Hz.  Note that the actual
+  pitch estimate is for the centre of the M sample Sn[] vector, not
+  the current N sample input vector.  This is (I think) a delay of 2.5
+  frames with N=80 samples.  You should align further analysis using
+  this pitch estimate to be centred on the middle of Sn[].
+
+  Two post processors have been tried, the MBE version (as discussed
+  in [1]), and a post processor that checks sub-multiples.  Both
+  suffer occasional gross pitch errors (i.e. neither are perfect).  In
+  the presence of background noise the sub-multiple algorithm tends
+  towards low F0 which leads to better sounding background noise than
+  the MBE post processor.
+
+  A good way to test and develop the NLP pitch estimator is using the
+  tnlp (codec2/unittest) and the codec2/octave/plnlp.m Octave script.
+
+  A pitch tracker searching a few frames forward and backward in time
+  would be a useful addition.
+
+  References:
+
+    [1] http://www.itr.unisa.edu.au/~steven/thesis/dgr.pdf Chapter 4
+                                                              
+\*---------------------------------------------------------------------------*/
+
+float nlp(
+  void *nlp_state, 
+  float  Sn[],			/* input speech vector */
+  int    n,			/* frames shift (no. new samples in Sn[]) */
+  int    m,			/* analysis window size */
+  int    pmin,			/* minimum pitch value */
+  int    pmax,			/* maximum pitch value */
+  float *pitch,			/* estimated pitch period in samples */
+  COMP   Sw[],                  /* Freq domain version of Sn[] */
+  float *prev_Wo
+)
+{
+    NLP   *nlp;
+    float  notch;		    /* current notch filter output */
+    COMP   Fw[PE_FFT_SIZE];	    /* DFT of squared signal */
+    float  gmax;
+    int    gmax_bin;
+    int   i,j;
+    float best_f0;
+
+    assert(nlp_state != NULL);
+    nlp = (NLP*)nlp_state;
+
+    /* Square, notch filter at DC, and LP filter vector */
+
+    for(i=m-n; i<M; i++) 	    /* square latest speech samples */
+	nlp->sq[i] = Sn[i]*Sn[i];
+
+    for(i=m-n; i<m; i++) {	/* notch filter at DC */
+	notch = nlp->sq[i] - nlp->mem_x;
+	notch += COEFF*nlp->mem_y;
+	nlp->mem_x = nlp->sq[i];
+	nlp->mem_y = notch;
+	nlp->sq[i] = notch;
+    }
+
+    for(i=m-n; i<m; i++) {	/* FIR filter vector */
+
+	for(j=0; j<NLP_NTAP-1; j++)
+	    nlp->mem_fir[j] = nlp->mem_fir[j+1];
+	nlp->mem_fir[NLP_NTAP-1] = nlp->sq[i];
+
+	nlp->sq[i] = 0.0;
+	for(j=0; j<NLP_NTAP; j++)
+	    nlp->sq[i] += nlp->mem_fir[j]*nlp_fir[j];
+    }
+
+    /* Decimate and DFT */
+
+    for(i=0; i<PE_FFT_SIZE; i++) {
+	Fw[i].real = 0.0;
+	Fw[i].imag = 0.0;
+    }
+    for(i=0; i<m/DEC; i++) {
+	Fw[i].real = nlp->sq[i*DEC]*(0.5 - 0.5*cos(2*PI*i/(m/DEC-1)));
+    }
+    dump_dec(Fw);
+    four1(&Fw[-1].imag,PE_FFT_SIZE,1);
+    for(i=0; i<PE_FFT_SIZE; i++)
+	Fw[i].real = Fw[i].real*Fw[i].real + Fw[i].imag*Fw[i].imag;
+
+    dump_sq(nlp->sq);
+    dump_Fw(Fw);
+
+    /* find global peak */
+
+    gmax = 0.0;
+    gmax_bin = PE_FFT_SIZE*DEC/pmax;
+    for(i=PE_FFT_SIZE*DEC/pmax; i<=PE_FFT_SIZE*DEC/pmin; i++) {
+	if (Fw[i].real > gmax) {
+	    gmax = Fw[i].real;
+	    gmax_bin = i;
+	}
+    }
+
+    best_f0 = post_process_sub_multiples(Fw, pmin, pmax, gmax, gmax_bin, 
+					 prev_Wo);
+
+    /* Shift samples in buffer to make room for new samples */
+
+    for(i=0; i<m-n; i++)
+	nlp->sq[i] = nlp->sq[i+n];
+
+    /* return pitch and F0 estimate */
+
+    *pitch = (float)SAMPLE_RATE/best_f0;
+    return(best_f0);  
+}
+
+/*---------------------------------------------------------------------------*\
+                                                                             
+  post_process_sub_multiples() 
+                                                                           
+  Given the global maximma of Fw[] we search interger submultiples for
+  local maxima.  If local maxima exist and they are above an
+  experimentally derived threshold (OK a magic number I pulled out of
+  the air) we choose the submultiple as the F0 estimate.
+
+  The rational for this is that the lowest frequency peak of Fw[]
+  should be F0, as Fw[] can be considered the autocorrelation function
+  of Sw[] (the speech spectrum).  However sometimes due to phase
+  effects the lowest frequency maxima may not be the global maxima.
+
+  This works OK in practice and favours low F0 values in the presence
+  of background noise which means the sinusoidal codec does an OK job
+  of synthesising the background noise.  High F0 in background noise
+  tends to sound more periodic introducing annoying artifacts.
+
+\*---------------------------------------------------------------------------*/
+
+float post_process_sub_multiples(COMP Fw[], 
+				 int pmin, int pmax, float gmax, int gmax_bin,
+				 float *prev_Wo)
+{
+    int   min_bin, cmax_bin;
+    int   mult;
+    float thresh, best_f0;
+    int   b, bmin, bmax, lmax_bin;
+    float lmax, cmax;
+    int   prev_f0_bin;
+
+    /* post process estimate by searching submultiples */
+
+    mult = 2;
+    min_bin = PE_FFT_SIZE*DEC/pmax;
+    cmax_bin = gmax_bin;
+    prev_f0_bin = *prev_Wo*(4000.0/PI)*(PE_FFT_SIZE*DEC)/SAMPLE_RATE;
+
+    while(gmax_bin/mult >= min_bin) {
+
+	b = gmax_bin/mult;			/* determine search interval */
+	bmin = 0.8*b;
+	bmax = 1.2*b;
+	if (bmin < min_bin)
+	    bmin = min_bin;
+
+	/* lower threshold to favour previous frames pitch estimate,
+	    this is a form of pitch tracking */
+
+	if ((prev_f0_bin > bmin) && (prev_f0_bin < bmax))
+	    thresh = CNLP*0.5*gmax;
+	else
+	    thresh = CNLP*gmax;
+
+	lmax = 0;
+	lmax_bin = bmin;
+	for (b=bmin; b<=bmax; b++) 		/* look for maximum in interval */
+	    if (Fw[b].real > lmax) {
+		lmax = Fw[b].real;
+		lmax_bin = b;
+	    }
+
+	if (lmax > thresh)
+	    if ((lmax > Fw[lmax_bin-1].real) && (lmax > Fw[lmax_bin+1].real)) {
+		cmax = lmax;
+		cmax_bin = lmax_bin;
+	    }
+
+	mult++;
+    }
+
+    best_f0 = (float)cmax_bin*SAMPLE_RATE/(PE_FFT_SIZE*DEC);
+
+    return best_f0;
+}
+  

libs/libcodec2/src/nlp.h

@@ -0,0 +1,38 @@
+/*---------------------------------------------------------------------------*\
+                                                 
+  FILE........: nlp.c                                                   
+  AUTHOR......: David Rowe                                      
+  DATE CREATED: 23/3/93                                    
+                                                         
+  Non Linear Pitch (NLP) estimation functions.			  
+                                                               
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#ifndef __NLP__
+#define __NLP__
+
+void *nlp_create();
+void nlp_destroy(void *nlp_state);
+float nlp(void *nlp_state, float Sn[], int n, int m, int pmin, int pmax, 
+	  float *pitch, COMP Sw[], float *prev_Wo);
+float test_candidate_mbe(COMP Sw[], float f0, COMP Sw_[]);
+
+#endif

libs/libcodec2/src/pack.c

@@ -0,0 +1,104 @@
+/*
+  Copyright (C) 2010 Perens LLC <bruce@perens.com>
+
+  This program 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, either version 3 of the License, or
+  (at your option) any later version.
+
+  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, see <http://www.gnu.org/licenses/>.
+
+ */
+#include "defines.h"
+#include "quantise.h"
+#include <stdio.h>
+
+/* Compile-time constants */
+/* Size of unsigned char in bits. Assumes 8 bits-per-char. */
+static const unsigned int	WordSize = 8;
+
+/* Mask to pick the bit component out of bitIndex. */
+static const unsigned int	IndexMask = 0x7;
+
+/* Used to pick the word component out of bitIndex. */
+static const unsigned int	ShiftRight = 3;
+
+/** Pack a bit field into a bit string, encoding the field in Gray code.
+ *
+ * The output is an array of unsigned char data. The fields are efficiently
+ * packed into the bit string. The Gray coding is a naive attempt to reduce
+ * the effect of single-bit errors, we expect to do a better job as the
+ * codec develops.
+ *
+ * This code would be simpler if it just set one bit at a time in the string,
+ * but would hit the same cache line more often. I'm not sure the complexity
+ * gains us anything here.
+ *
+ * Although field is currently of int type rather than unsigned for
+ * compatibility with the rest of the code, indices are always expected to
+ * be >= 0.
+ */
+void
+pack(
+ unsigned char *	bitArray, /* The output bit string. */
+ unsigned int *		bitIndex, /* Index into the string in BITS, not bytes.*/
+ int			field,	  /* The bit field to be packed. */
+ unsigned int		fieldWidth/* Width of the field in BITS, not bytes. */
+ )
+{
+  /* Convert the field to Gray code */
+  field = (field >> 1) ^ field;
+
+  do {
+    unsigned int  	bI = *bitIndex;
+    unsigned int	bitsLeft = WordSize - (bI & IndexMask);
+    unsigned int	sliceWidth =
+			 bitsLeft < fieldWidth ? bitsLeft : fieldWidth;
+    unsigned int	wordIndex = bI >> ShiftRight;
+
+    bitArray[wordIndex] |=
+     ((unsigned char)((field >> (fieldWidth - sliceWidth))
+     << (bitsLeft - sliceWidth)));
+    
+    *bitIndex = bI + sliceWidth;
+    fieldWidth -= sliceWidth;
+  } while ( fieldWidth != 0 );
+}
+
+/** Unpack a field from a bit string, converting from Gray code to binary.
+ *
+ */
+int
+unpack(
+ const unsigned char *	bitArray, /* The input bit string. */
+ unsigned int *		bitIndex, /* Index into the string in BITS, not bytes.*/
+ unsigned int		fieldWidth/* Width of the field in BITS, not bytes. */
+ )
+{
+  unsigned int		field = 0;
+
+  do {
+    unsigned int  	bI = *bitIndex;
+    unsigned int	bitsLeft = WordSize - (bI & IndexMask);
+    unsigned int	sliceWidth =
+			 bitsLeft < fieldWidth ? bitsLeft : fieldWidth;
+
+    field |= (((bitArray[bI >> ShiftRight] >> (bitsLeft - sliceWidth)) & ((1 << sliceWidth) - 1)) << (fieldWidth - sliceWidth));
+    
+    *bitIndex = bI + sliceWidth;
+    fieldWidth -= sliceWidth;
+  } while ( fieldWidth != 0 );
+
+  /* Convert from Gray code to binary. Works for maximum 8-bit fields. */
+  unsigned int	t = field ^ (field >> 8);
+  t ^= (t >> 4);
+  t ^= (t >> 2);
+  t ^= (t >> 1);
+  return t;
+}

libs/libcodec2/src/phase.c

@@ -0,0 +1,254 @@
+/*---------------------------------------------------------------------------*\
+                                                                             
+  FILE........: phase.c                                           
+  AUTHOR......: David Rowe                                             
+  DATE CREATED: 1/2/09                                                 
+                                                                             
+  Functions for modelling and synthesising phase.
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#include "defines.h"
+#include "phase.h"
+#include "four1.h"
+
+#include <assert.h>
+#include <math.h>
+#include <string.h>
+#include <stdlib.h>
+
+#define VTHRESH 4.0
+
+/*---------------------------------------------------------------------------*\
+
+  aks_to_H()
+
+  Samples the complex LPC synthesis filter spectrum at the harmonic
+  frequencies.
+
+\*---------------------------------------------------------------------------*/
+
+void aks_to_H(
+	      MODEL *model,	/* model parameters */
+	      float  aks[],	/* LPC's */
+	      float  G,	        /* energy term */
+	      COMP   H[],	/* complex LPC spectral samples */
+	      int    order
+)
+{
+  COMP  Pw[FFT_DEC];	/* power spectrum */
+  int   i,m;		/* loop variables */
+  int   am,bm;		/* limits of current band */
+  float r;		/* no. rads/bin */
+  float Em;		/* energy in band */
+  float Am;		/* spectral amplitude sample */
+  int   b;		/* centre bin of harmonic */
+  float phi_;		/* phase of LPC spectra */
+
+  r = TWO_PI/(FFT_DEC);
+
+  /* Determine DFT of A(exp(jw)) ------------------------------------------*/
+
+  for(i=0; i<FFT_DEC; i++) {
+    Pw[i].real = 0.0;
+    Pw[i].imag = 0.0;
+  }
+
+  for(i=0; i<=order; i++)
+    Pw[i].real = aks[i];
+
+  four1(&Pw[-1].imag,FFT_DEC,-1);
+
+  /* Sample magnitude and phase at harmonics */
+
+  for(m=1; m<=model->L; m++) {
+    am = floor((m - 0.5)*model->Wo/r + 0.5);
+    bm = floor((m + 0.5)*model->Wo/r + 0.5);
+    b = floor(m*model->Wo/r + 0.5);
+
+    Em = 0.0;
+    for(i=am; i<bm; i++)
+      Em += G/(Pw[i].real*Pw[i].real + Pw[i].imag*Pw[i].imag);
+    Am = sqrt(fabs(Em/(bm-am)));
+
+    phi_ = -atan2(Pw[b].imag,Pw[b].real);
+    H[m].real = Am*cos(phi_);
+    H[m].imag = Am*sin(phi_);
+  }
+}
+
+
+/*---------------------------------------------------------------------------*\
+
+   phase_synth_zero_order()
+
+   Synthesises phases based on SNR and a rule based approach.  No phase 
+   parameters are required apart from the SNR (which can be reduced to a
+   1 bit V/UV decision per frame).
+
+   The phase of each harmonic is modelled as the phase of a LPC
+   synthesis filter excited by an impulse.  Unlike the first order
+   model the position of the impulse is not transmitted, so we create
+   an excitation pulse train using a rule based approach.  
+
+   Consider a pulse train with a pulse starting time n=0, with pulses
+   repeated at a rate of Wo, the fundamental frequency.  A pulse train
+   in the time domain is equivalent to harmonics in the frequency
+   domain.  We can make an excitation pulse train using a sum of
+   sinsusoids:
+
+     for(m=1; m<=L; m++)
+       ex[n] = cos(m*Wo*n)
+
+   Note: the Octave script ../octave/phase.m is an example of this if
+   you would like to try making a pulse train.
+
+   The phase of each excitation harmonic is:
+
+     arg(E[m]) = mWo
+
+   where E[m] are the complex excitation (freq domain) samples,
+   arg(x), just returns the phase of a complex sample x.
+
+   As we don't transmit the pulse position for this model, we need to
+   synthesise it.  Now the excitation pulses occur at a rate of Wo.
+   This means the phase of the first harmonic advances by N samples
+   over a synthesis frame of N samples.  For example if Wo is pi/20
+   (200 Hz), then over a 10ms frame (N=80 samples), the phase of the
+   first harmonic would advance (pi/20)*80 = 4*pi or two complete
+   cycles.
+
+   We generate the excitation phase of the fundamental (first
+   harmonic):
+
+     arg[E[1]] = Wo*N;
+
+   We then relate the phase of the m-th excitation harmonic to the
+   phase of the fundamental as:
+
+     arg(E[m]) = m*arg(E[1])
+
+   This E[m] then gets passed through the LPC synthesis filter to
+   determine the final harmonic phase.
+     
+   For a while there were prolems with low pitched males like hts1
+   sounding "clicky".  The synthesied time domain waveform also looked
+   clicky.  Many methods were tried to improve the sounds quality of
+   low pitched males. Finally adding a small amount of jitter to each
+   harmonic worked.
+
+   The current result sounds very close to the original phases, with
+   only 1 voicing bit per frame.  For example hts1a using original
+   amplitudes and this phase model produces speech hard to distinguish
+   from speech synthesise with the orginal phases.  The sound quality
+   of this patrtiallyuantised codec (nb original amplitudes) is higher
+   than g729, even though all the phase information has been
+   discarded.
+
+   NOTES:
+
+     1/ This synthesis model is effectvely the same as simple LPC-10
+     vocoders, and yet sounds much better.  Why?
+
+     2/ I am pretty sure the Lincoln Lab sinusoidal coding guys (like xMBE
+     also from MIT) first described this zero phase model, I need to look
+     up the paper.
+
+     3/ Note that this approach could cause some discontinuities in
+     the phase at the edge of synthesis frames, as no attempt is made
+     to make sure that the phase tracks are continuous (the excitation
+     phases are continuous, but not the final phases after filtering
+     by the LPC spectra).  Technically this is a bad thing.  However
+     this may actually be a good thing, disturbing the phase tracks a
+     bit.  More research needed, e.g. test a synthesis model that adds
+     a small delta-W to make phase tracks line up for voiced
+     harmonics.
+
+     4/ Why does this sound so great with 1 V/UV decision?  Conventional
+     wisdom says mixed voicing is required for high qaulity speech.
+
+\*---------------------------------------------------------------------------*/
+
+void phase_synth_zero_order(
+    MODEL *model,
+    float  aks[],
+    float *ex_phase             /* excitation phase of fundamental */
+)
+{
+  int   m;
+  float new_phi;
+  COMP  Ex[MAX_AMP];		/* excitation samples */
+  COMP  A_[MAX_AMP];		/* synthesised harmonic samples */
+  COMP  H[MAX_AMP];             /* LPC freq domain samples */
+  float G;
+  float jitter;
+
+  G = 1.0;
+  aks_to_H(model,aks,G,H,LPC_ORD);
+
+  /* 
+     Update excitation fundamental phase track, this sets the position
+     of each pitch pulse during voiced speech.  After much experiment
+     I found that using just this frame Wo improved quality for UV
+     sounds compared to interpolating two frames Wo like this:
+     
+     ex_phase[0] += (*prev_Wo+mode->Wo)*N/2;
+  */
+  
+  ex_phase[0] += (model->Wo)*N;
+  ex_phase[0] -= TWO_PI*floor(ex_phase[0]/TWO_PI + 0.5);
+
+  for(m=1; m<=model->L; m++) {
+
+    /* generate excitation */
+
+    if (model->voiced) {
+	/* This method of adding jitter really helped remove the clicky
+	   sound in low pitched makes like hts1a. This moves the onset
+	   of each harmonic over at +/- 0.25 of a sample.
+	*/
+        jitter = 0.25*(1.0 - 2.0*rand()/RAND_MAX);
+	Ex[m].real = cos(ex_phase[0]*m - jitter*model->Wo*m);
+	Ex[m].imag = sin(ex_phase[0]*m - jitter*model->Wo*m);
+    }
+    else {
+
+	/* When a few samples were tested I found that LPC filter
+	   phase is not needed in the unvoiced case, but no harm in
+	   keeping it.
+        */
+	float phi = TWO_PI*(float)rand()/RAND_MAX;
+        Ex[m].real = cos(phi);
+	Ex[m].imag = sin(phi);
+    }
+
+    /* filter using LPC filter */
+
+    A_[m].real = H[m].real*Ex[m].real - H[m].imag*Ex[m].imag;
+    A_[m].imag = H[m].imag*Ex[m].real + H[m].real*Ex[m].imag;
+
+    /* modify sinusoidal phase */
+   
+    new_phi = atan2(A_[m].imag, A_[m].real+1E-12);
+    model->phi[m] = new_phi;
+  }
+
+}

libs/libcodec2/src/phase.h

@@ -0,0 +1,34 @@
+/*---------------------------------------------------------------------------*\
+                                                                             
+  FILE........: phase.h                                          
+  AUTHOR......: David Rowe                                             
+  DATE CREATED: 1/2/09                                                 
+                                                                             
+  Functions for modelling phase.
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#ifndef __PHASE__
+#define __PHASE__
+
+void phase_synth_zero_order(MODEL *model, float aks[], float *ex_phase);
+
+#endif

libs/libcodec2/src/postfilter.c

@@ -0,0 +1,131 @@
+/*---------------------------------------------------------------------------*\
+                                                                             
+  FILE........: postfilter.c
+  AUTHOR......: David Rowe                                                          
+  DATE CREATED: 13/09/09
+                                                                             
+  Postfilter to improve sound quality for speech with high levels of
+  background noise.  Unlike mixed-excitation models requires no bits
+  to be transmitted to handle background noise.
+                                                                             
+\*---------------------------------------------------------------------------*/
+
+/*
+  Copyright (C) 2009 David Rowe
+
+  All rights reserved.
+
+  This program is free software; you can redistribute it and/or modify
+  it under the terms of the GNU Lesser 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 Lesser General Public License
+  along with this program; if not, write to the Free Software
+  Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
+*/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <math.h>
+
+#include "defines.h"
+#include "dump.h"
+#include "postfilter.h"
+
+/*---------------------------------------------------------------------------*\
+
+                                DEFINES
+
+\*---------------------------------------------------------------------------*/
+
+#define BG_THRESH 40.0     /* only consider low levels signals for bg_est */
+#define BG_BETA    0.1     /* averaging filter constant                   */
+
+/*---------------------------------------------------------------------------*\
+
+  postfilter()
+
+  The post filter is designed to help with speech corrupted by
+  background noise.  The zero phase model tends to make speech with
+  background noise sound "clicky".  With high levels of background
+  noise the low level inter-formant parts of the spectrum will contain
+  noise rather than speech harmonics, so modelling them as voiced
+  (i.e. a continuous, non-random phase track) is inaccurate.
+
+  Some codecs (like MBE) have a mixed voicing model that breaks the
+  spectrum into voiced and unvoiced regions.  Several bits/frame
+  (5-12) are required to transmit the frequency selective voicing
+  information.  Mixed excitation also requires accurate voicing
+  estimation (parameter estimators always break occasionally under
+  exceptional condition).
+
+  In our case we use a post filter approach which requires no
+  additional bits to be transmitted.  The decoder measures the average
+  level of the background noise during unvoiced frames.  If a harmonic
+  is less than this level it is made unvoiced by randomising it's
+  phases.
+
+  This idea is rather experimental.  Some potential problems that may
+  happen:
+ 
+  1/ If someone says "aaaaaaaahhhhhhhhh" will background estimator track
+     up to speech level?  This would be a bad thing.
+
+  2/ If background noise suddenly dissapears from the source speech does
+     estimate drop quickly?  What is noise suddenly re-appears?
+
+  3/ Background noise with a non-flat sepctrum.  Current algorithm just
+     comsiders scpetrum as a whole, but this could be broken up into
+     bands, each with their own estimator.
+
+  4/ Males and females with the same level of background noise.  Check
+     performance the same.  Changing Wo affects width of each band, may
+     affect bg energy estimates.
+
+  5/ Not sure what happens during long periods of voiced speech
+     e.g. "sshhhhhhh"
+  
+\*---------------------------------------------------------------------------*/
+
+void postfilter(
+  MODEL *model,
+  float *bg_est
+)	
+{
+  int   m, uv;
+  float e;
+
+  /* determine average energy across spectrum */
+
+  e = 0.0;
+  for(m=1; m<=model->L; m++)
+      e += model->A[m]*model->A[m];
+
+  e = 10.0*log10(e/model->L);
+
+  /* If beneath threhold, update bg estimate.  The idea
+     of the threshold is to prevent updating during high level
+     speech. */
+
+  if ((e < BG_THRESH) && !model->voiced)
+      *bg_est =  *bg_est*(1.0 - BG_BETA) + e*BG_BETA;
+
+  /* now mess with phases during voiced frames to make any harmonics
+     less then our background estimate unvoiced.
+  */
+
+  uv = 0;
+  if (model->voiced)
+      for(m=1; m<=model->L; m++)
+	  if (20.0*log10(model->A[m]) < *bg_est) {
+	      model->phi[m] = TWO_PI*(float)rand()/RAND_MAX;
+	      uv++;
+	  }
+
+  dump_bg(e, *bg_est, 100.0*uv/model->L);
+
+}