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
AC_CONFIG_SUBDIRS([libs/broadvoice])
AC_CONFIG_SUBDIRS([libs/libg722_1])
AC_CONFIG_SUBDIRS([libs/silk])
+AC_CONFIG_SUBDIRS([libs/libcodec2])
case $host in
*-openbsd*)
--- /dev/null
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--- /dev/null
+Installation Instructions
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+ If you have to use a `make' that does not support the `VPATH'
+variable, you have to compile the package for one architecture at a
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+ if ($(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
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+ if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
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+ else \
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+ fi; \
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+ ! -type d ! -perm -444 -exec $(SHELL) $(install_sh) -c -m a+r {} {} \; \
+ || chmod -R a+r $(distdir)
+dist-gzip: distdir
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+
+# This target untars the dist file and tries a VPATH configuration. Then
+# it guarantees that the distribution is self-contained by making another
+# tarfile.
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+ *.tar.gz*) \
+ GZIP=$(GZIP_ENV) gunzip -c $(distdir).tar.gz | $(am__untar) ;;\
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+ *.shar.gz*) \
+ GZIP=$(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 $(distdir)/_inst
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+ dc_install_base=`$(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 \
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+ && $(MAKE) $(AM_MAKEFLAGS) DESTDIR="$$dc_destdir" \
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+ && $(MAKE) $(AM_MAKEFLAGS) dist \
+ && rm -rf $(DIST_ARCHIVES) \
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+ $(am__remove_distdir)
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+distuninstallcheck:
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+ && 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
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+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:
--- /dev/null
+#! /bin/sh
+srcpath=$(dirname $0 2>/dev/null ) || srcpath="."
+$srcpath/configure "$@" --disable-shared --with-pic
+
--- /dev/null
+# -*- 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
--- /dev/null
+s(n)=A_1cos(\omega_0+\phi_1)+A_2cos(2\omega_0+\phi_2)+...+A_Lcos(L\omega_0+\phi_L)
--- /dev/null
+% 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);
--- /dev/null
+% 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
--- /dev/null
+% 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
--- /dev/null
+% 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
+
--- /dev/null
+% 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
+
--- /dev/null
+% 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
+
--- /dev/null
+% 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
--- /dev/null
+% 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
--- /dev/null
+% 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
--- /dev/null
+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])
--- /dev/null
+% 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
--- /dev/null
+% 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
--- /dev/null
+% 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
+
--- /dev/null
+% 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
--- /dev/null
+% 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
+
--- /dev/null
+% 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
+
--- /dev/null
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--- /dev/null
+#!/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
--- /dev/null
+#!/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
--- /dev/null
+#!/bin/sh
+# Converts 16 bit signed short 8 kHz raw (headerless) files to wave
+sox -r 8000 -s -2 $1 $2
--- /dev/null
+#!/bin/sh
+# Converts wave files to raw (headerless) files
+sox $1 -t raw $2
--- /dev/null
+AM_CFLAGS = -I../src -Wall -DFLOATING_POINT -DVAR_ARRAYS
+AUTOMAKE_OPTS = gnu
+NAME = libcodec2
+AM_CPPFLAGS = $(AM_CFLAGS)
+
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+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
+
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+libcodec2_la_LDFLAGS = $(LIBS)
+
+library_includedir = $(prefix)
+library_include_HEADERS = codec2.h \
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+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)
--- /dev/null
+# 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
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+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
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+am__aclocal_m4_deps = $(top_srcdir)/configure.in
+am__configure_deps = $(am__aclocal_m4_deps) $(CONFIGURE_DEPENDENCIES) \
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+ libcodec2_la-four1.lo libcodec2_la-interp.lo \
+ libcodec2_la-lsp.lo libcodec2_la-phase.lo \
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+ $(AM_LDFLAGS) $(LDFLAGS) -o $@
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+# 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:
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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;
+}
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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;
+}
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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];
+ }
+
+}
--- /dev/null
+#!/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
--- /dev/null
+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
+};
+
--- /dev/null
+#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
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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_);
+}
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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));
+}
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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__ */
+
--- /dev/null
+#!/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
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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 */
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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 */
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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;
+}
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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
--- /dev/null
+#!/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
+
+
--- /dev/null
+#!/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
+
+
--- /dev/null
+#!/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
+
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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);
+}
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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
--- /dev/null
+/*---------------------------------------------------------------------------*\\r
+\r
+ FILE........: lsp.c\r
+ AUTHOR......: David Rowe\r
+ DATE CREATED: 24/2/93\r
+\r
+\r
+ This file contains functions for LPC to LSP conversion and LSP to\r
+ LPC conversion. Note that the LSP coefficients are not in radians\r
+ format but in the x domain of the unit circle.\r
+\r
+\*---------------------------------------------------------------------------*/\r
+\r
+#include "defines.h"\r
+#include "lsp.h"\r
+#include <math.h>\r
+#include <stdio.h>\r
+#include <stdlib.h>\r
+\r
+/*---------------------------------------------------------------------------*\\r
+\r
+ Introduction to Line Spectrum Pairs (LSPs)\r
+ ------------------------------------------\r
+\r
+ LSPs are used to encode the LPC filter coefficients {ak} for\r
+ transmission over the channel. LSPs have several properties (like\r
+ less sensitivity to quantisation noise) that make them superior to\r
+ direct quantisation of {ak}.\r
+\r
+ A(z) is a polynomial of order lpcrdr with {ak} as the coefficients.\r
+\r
+ A(z) is transformed to P(z) and Q(z) (using a substitution and some\r
+ algebra), to obtain something like:\r
+\r
+ A(z) = 0.5[P(z)(z+z^-1) + Q(z)(z-z^-1)] (1)\r
+\r
+ As you can imagine A(z) has complex zeros all over the z-plane. P(z)\r
+ and Q(z) have the very neat property of only having zeros _on_ the\r
+ unit circle. So to find them we take a test point z=exp(jw) and\r
+ evaluate P (exp(jw)) and Q(exp(jw)) using a grid of points between 0\r
+ and pi.\r
+\r
+ The zeros (roots) of P(z) also happen to alternate, which is why we\r
+ swap coefficients as we find roots. So the process of finding the\r
+ LSP frequencies is basically finding the roots of 5th order\r
+ polynomials.\r
+\r
+ The root so P(z) and Q(z) occur in symmetrical pairs at +/-w, hence\r
+ the name Line Spectrum Pairs (LSPs).\r
+\r
+ To convert back to ak we just evaluate (1), "clocking" an impulse\r
+ thru it lpcrdr times gives us the impulse response of A(z) which is\r
+ {ak}.\r
+\r
+\*---------------------------------------------------------------------------*/\r
+\r
+/*---------------------------------------------------------------------------*\\r
+\r
+ FUNCTION....: cheb_poly_eva()\r
+ AUTHOR......: David Rowe\r
+ DATE CREATED: 24/2/93\r
+\r
+ This function evalutes a series of chebyshev polynomials\r
+\r
+ FIXME: performing memory allocation at run time is very inefficient,\r
+ replace with stack variables of MAX_P size.\r
+\r
+\*---------------------------------------------------------------------------*/\r
+\r
+\r
+float cheb_poly_eva(float *coef,float x,int m)\r
+/* float coef[] coefficients of the polynomial to be evaluated */\r
+/* float x the point where polynomial is to be evaluated */\r
+/* int m order of the polynomial */\r
+{\r
+ int i;\r
+ float *T,*t,*u,*v,sum;\r
+\r
+ /* Allocate memory for chebyshev series formulation */\r
+\r
+ if((T = (float *)malloc((m/2+1)*sizeof(float))) == NULL){\r
+ fprintf(stderr, "not enough memory to allocate buffer\n");\r
+ exit(1);\r
+ }\r
+\r
+ /* Initialise pointers */\r
+\r
+ t = T; /* T[i-2] */\r
+ *t++ = 1.0;\r
+ u = t--; /* T[i-1] */\r
+ *u++ = x;\r
+ v = u--; /* T[i] */\r
+\r
+ /* Evaluate chebyshev series formulation using iterative approach */\r
+\r
+ for(i=2;i<=m/2;i++)\r
+ *v++ = (2*x)*(*u++) - *t++; /* T[i] = 2*x*T[i-1] - T[i-2] */\r
+\r
+ sum=0.0; /* initialise sum to zero */\r
+ t = T; /* reset pointer */\r
+\r
+ /* Evaluate polynomial and return value also free memory space */\r
+\r
+ for(i=0;i<=m/2;i++)\r
+ sum+=coef[(m/2)-i]**t++;\r
+\r
+ free(T);\r
+ return sum;\r
+}\r
+\r
+\r
+/*---------------------------------------------------------------------------*\\r
+\r
+ FUNCTION....: lpc_to_lsp()\r
+ AUTHOR......: David Rowe\r
+ DATE CREATED: 24/2/93\r
+\r
+ This function converts LPC coefficients to LSP coefficients.\r
+\r
+\*---------------------------------------------------------------------------*/\r
+\r
+int lpc_to_lsp (float *a, int lpcrdr, float *freq, int nb, float delta)\r
+/* float *a lpc coefficients */\r
+/* int lpcrdr order of LPC coefficients (10) */\r
+/* float *freq LSP frequencies in radians */\r
+/* int nb number of sub-intervals (4) */\r
+/* float delta grid spacing interval (0.02) */\r
+{\r
+ float psuml,psumr,psumm,temp_xr,xl,xr,xm;\r
+ float temp_psumr;\r
+ int i,j,m,flag,k;\r
+ float *Q; /* ptrs for memory allocation */\r
+ float *P;\r
+ float *px; /* ptrs of respective P'(z) & Q'(z) */\r
+ float *qx;\r
+ float *p;\r
+ float *q;\r
+ float *pt; /* ptr used for cheb_poly_eval()\r
+ whether P' or Q' */\r
+ int roots=0; /* number of roots found */\r
+ flag = 1; \r
+ m = lpcrdr/2; /* order of P'(z) & Q'(z) polynimials */\r
+\r
+ /* Allocate memory space for polynomials */\r
+\r
+ Q = (float *) malloc((m+1)*sizeof(float));\r
+ P = (float *) malloc((m+1)*sizeof(float));\r
+ if( (P == NULL) || (Q == NULL) ) {\r
+ fprintf(stderr,"not enough memory to allocate buffer\n");\r
+ exit(1);\r
+ }\r
+\r
+ /* determine P'(z)'s and Q'(z)'s coefficients where\r
+ P'(z) = P(z)/(1 + z^(-1)) and Q'(z) = Q(z)/(1-z^(-1)) */\r
+\r
+ px = P; /* initilaise ptrs */\r
+ qx = Q;\r
+ p = px;\r
+ q = qx;\r
+ *px++ = 1.0;\r
+ *qx++ = 1.0;\r
+ for(i=1;i<=m;i++){\r
+ *px++ = a[i]+a[lpcrdr+1-i]-*p++;\r
+ *qx++ = a[i]-a[lpcrdr+1-i]+*q++;\r
+ }\r
+ px = P;\r
+ qx = Q;\r
+ for(i=0;i<m;i++){\r
+ *px = 2**px;\r
+ *qx = 2**qx;\r
+ px++;\r
+ qx++;\r
+ }\r
+ px = P; /* re-initialise ptrs */\r
+ qx = Q;\r
+\r
+ /* Search for a zero in P'(z) polynomial first and then alternate to Q'(z).\r
+ Keep alternating between the two polynomials as each zero is found */\r
+\r
+ xr = 0; /* initialise xr to zero */\r
+ xl = 1.0; /* start at point xl = 1 */\r
+\r
+\r
+ for(j=0;j<lpcrdr;j++){\r
+ if(j%2) /* determines whether P' or Q' is eval. */\r
+ pt = qx;\r
+ else\r
+ pt = px;\r
+\r
+ psuml = cheb_poly_eva(pt,xl,lpcrdr); /* evals poly. at xl */\r
+ flag = 1;\r
+ while(flag && (xr >= -1.0)){\r
+ xr = xl - delta ; /* interval spacing */\r
+ psumr = cheb_poly_eva(pt,xr,lpcrdr);/* poly(xl-delta_x) */\r
+ temp_psumr = psumr;\r
+ temp_xr = xr;\r
+\r
+ /* if no sign change increment xr and re-evaluate\r
+ poly(xr). Repeat til sign change. if a sign change has\r
+ occurred the interval is bisected and then checked again\r
+ for a sign change which determines in which interval the\r
+ zero lies in. If there is no sign change between poly(xm)\r
+ and poly(xl) set interval between xm and xr else set\r
+ interval between xl and xr and repeat till root is located\r
+ within the specified limits */\r
+\r
+ if((psumr*psuml)<0.0){\r
+ roots++;\r
+\r
+ psumm=psuml;\r
+ for(k=0;k<=nb;k++){\r
+ xm = (xl+xr)/2; /* bisect the interval */\r
+ psumm=cheb_poly_eva(pt,xm,lpcrdr);\r
+ if(psumm*psuml>0.){\r
+ psuml=psumm;\r
+ xl=xm;\r
+ }\r
+ else{\r
+ psumr=psumm;\r
+ xr=xm;\r
+ }\r
+ }\r
+\r
+ /* once zero is found, reset initial interval to xr */\r
+ freq[j] = (xm);\r
+ xl = xm;\r
+ flag = 0; /* reset flag for next search */\r
+ }\r
+ else{\r
+ psuml=temp_psumr;\r
+ xl=temp_xr;\r
+ }\r
+ }\r
+ }\r
+ free(P); /* free memory space */\r
+ free(Q);\r
+\r
+ /* convert from x domain to radians */\r
+\r
+ for(i=0; i<lpcrdr; i++) {\r
+ freq[i] = acos(freq[i]);\r
+ }\r
+\r
+ return(roots);\r
+}\r
+\r
+/*---------------------------------------------------------------------------*\\r
+\r
+ FUNCTION....: lsp_to_lpc()\r
+ AUTHOR......: David Rowe\r
+ DATE CREATED: 24/2/93\r
+\r
+ This function converts LSP coefficients to LPC coefficients. In the\r
+ Speex code we worked out a wayto simplify this significantly.\r
+\r
+\*---------------------------------------------------------------------------*/\r
+\r
+void lsp_to_lpc(float *freq, float *ak, int lpcrdr)\r
+/* float *freq array of LSP frequencies in radians */\r
+/* float *ak array of LPC coefficients */\r
+/* int lpcrdr order of LPC coefficients */\r
+\r
+\r
+{\r
+ int i,j;\r
+ float xout1,xout2,xin1,xin2;\r
+ float *Wp;\r
+ float *pw,*n1,*n2,*n3,*n4;\r
+ int m = lpcrdr/2;\r
+\r
+ /* convert from radians to the x=cos(w) domain */\r
+\r
+ for(i=0; i<lpcrdr; i++)\r
+ freq[i] = cos(freq[i]);\r
+\r
+ if((Wp = (float *) malloc((4*m+2)*sizeof(float))) == NULL){\r
+ printf("not enough memory to allocate buffer\n");\r
+ exit(1);\r
+ }\r
+ pw = Wp;\r
+\r
+ /* initialise contents of array */\r
+\r
+ for(i=0;i<=4*m+1;i++){ /* set contents of buffer to 0 */\r
+ *pw++ = 0.0;\r
+ }\r
+\r
+ /* Set pointers up */\r
+\r
+ pw = Wp;\r
+ xin1 = 1.0;\r
+ xin2 = 1.0;\r
+\r
+ /* reconstruct P(z) and Q(z) by cascading second order polynomials\r
+ in form 1 - 2xz(-1) +z(-2), where x is the LSP coefficient */\r
+\r
+ for(j=0;j<=lpcrdr;j++){\r
+ for(i=0;i<m;i++){\r
+ n1 = pw+(i*4);\r
+ n2 = n1 + 1;\r
+ n3 = n2 + 1;\r
+ n4 = n3 + 1;\r
+ xout1 = xin1 - 2*(freq[2*i]) * *n1 + *n2;\r
+ xout2 = xin2 - 2*(freq[2*i+1]) * *n3 + *n4;\r
+ *n2 = *n1;\r
+ *n4 = *n3;\r
+ *n1 = xin1;\r
+ *n3 = xin2;\r
+ xin1 = xout1;\r
+ xin2 = xout2;\r
+ }\r
+ xout1 = xin1 + *(n4+1);\r
+ xout2 = xin2 - *(n4+2);\r
+ ak[j] = (xout1 + xout2)*0.5;\r
+ *(n4+1) = xin1;\r
+ *(n4+2) = xin2;\r
+\r
+ xin1 = 0.0;\r
+ xin2 = 0.0;\r
+ }\r
+ free(Wp);\r
+}\r
+\r
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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;
+}
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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
--- /dev/null
+/*
+ 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;
+}
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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;
+ }
+
+}
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ 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);
+
+}
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: postfilter.h
+ AUTHOR......: David Rowe
+ DATE CREATED: 13/09/09
+
+ Postfilter header file.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+ 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 __POSTFILTER__
+#define __POSTFILTER__
+
+void postfilter(MODEL *model, float *bg_est);
+
+#endif
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: quantise.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 31/5/92
+
+ Quantisation functions for the sinusoidal coder.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+ 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 <ctype.h>
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+
+#include "defines.h"
+#include "dump.h"
+#include "quantise.h"
+#include "lpc.h"
+#include "lsp.h"
+#include "four1.h"
+#include "codebook.h"
+
+#define LSP_DELTA1 0.01 /* grid spacing for LSP root searches */
+#define MAX_CB 20 /* max number of codebooks */
+
+/* describes each codebook */
+
+typedef struct {
+ int k; /* dimension of vector */
+ int log2m; /* number of bits in m */
+ int m; /* elements in codebook */
+ float *fn; /* file name of text file storing the VQ */
+} LSP_CB;
+
+/* lsp_q describes entire quantiser made up of several codebooks */
+
+#ifdef OLDER
+/* 10+10+6+6 = 32 bit LSP difference split VQ */
+
+LSP_CB lsp_q[] = {
+ {3, 1024, "/usr/src/freeswitch/libs/libcodec2-1.0/unittest/lspd123.txt"},
+ {3, 1024, "/usr/src/freeswitch/libs/libcodec2-1.0/unittest/lspd456.txt"},
+ {2, 64, "/usr/src/freeswitch/libs/libcodec2-1.0/unittest/lspd78.txt"},
+ {2, 64, "/usr/src/freeswitch/libs/libcodec2-1.0/unittest/lspd910.txt"},
+ {0, 0, ""}
+};
+#endif
+
+LSP_CB lsp_q[] = {
+ {1,4,16, codebook_lsp1 },
+ {1,4,16, codebook_lsp2 },
+ {1,4,16, codebook_lsp3 },
+ {1,4,16, codebook_lsp4 },
+ {1,4,16, codebook_lsp5 },
+ {1,4,16, codebook_lsp6 },
+ {1,4,16, codebook_lsp7 },
+ {1,3,8, codebook_lsp8 },
+ {1,3,8, codebook_lsp9 },
+ {1,2,4, codebook_lsp10 },
+ {0,0,0, NULL },
+};
+
+/* ptr to each codebook */
+
+static float *plsp_cb[MAX_CB];
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION HEADERS
+
+\*---------------------------------------------------------------------------*/
+
+float speech_to_uq_lsps(float lsp[], float ak[], float Sn[], float w[],
+ int order);
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTIONS
+
+\*---------------------------------------------------------------------------*/
+
+int lsp_bits(int i) {
+ return lsp_q[i].log2m;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ quantise_uniform
+
+ Simulates uniform quantising of a float.
+
+\*---------------------------------------------------------------------------*/
+
+void quantise_uniform(float *val, float min, float max, int bits)
+{
+ int levels = 1 << (bits-1);
+ float norm;
+ int index;
+
+ /* hard limit to quantiser range */
+
+ printf("min: %f max: %f val: %f ", min, max, val[0]);
+ if (val[0] < min) val[0] = min;
+ if (val[0] > max) val[0] = max;
+
+ norm = (*val - min)/(max-min);
+ printf("%f norm: %f ", val[0], norm);
+ index = fabs(levels*norm + 0.5);
+
+ *val = min + index*(max-min)/levels;
+
+ printf("index %d val_: %f\n", index, val[0]);
+}
+
+/*---------------------------------------------------------------------------*\
+
+ lspd_quantise
+
+ Simulates differential lsp quantiser
+
+\*---------------------------------------------------------------------------*/
+
+void lsp_quantise(
+ float lsp[],
+ float lsp_[],
+ int order
+)
+{
+ int i;
+ float dlsp[LPC_MAX];
+ float dlsp_[LPC_MAX];
+
+ dlsp[0] = lsp[0];
+ for(i=1; i<order; i++)
+ dlsp[i] = lsp[i] - lsp[i-1];
+
+ for(i=0; i<order; i++)
+ dlsp_[i] = dlsp[i];
+
+ quantise_uniform(&dlsp_[0], 0.1, 0.5, 5);
+
+ lsp_[0] = dlsp_[0];
+ for(i=1; i<order; i++)
+ lsp_[i] = lsp_[i-1] + dlsp_[i];
+}
+
+/*---------------------------------------------------------------------------*\
+
+ scan_line()
+
+ This function reads a vector of floats from a line in a text file.
+
+\*---------------------------------------------------------------------------*/
+
+void scan_line(FILE *fp, float f[], int n)
+/* FILE *fp; file ptr to text file */
+/* float f[]; array of floats to return */
+/* int n; number of floats in line */
+{
+ char s[MAX_STR];
+ char *ps,*pe;
+ int i;
+
+ fgets(s,MAX_STR,fp);
+ ps = pe = s;
+ for(i=0; i<n; i++) {
+ while( isspace(*pe)) pe++;
+ while( !isspace(*pe)) pe++;
+ sscanf(ps,"%f",&f[i]);
+ ps = pe;
+ }
+}
+
+/*---------------------------------------------------------------------------*\
+
+ load_cb
+
+ Loads a single codebook (LSP vector quantiser) into memory.
+
+\*---------------------------------------------------------------------------*/
+
+void load_cb(float *source, float *cb, int k, int m)
+{
+ int lines;
+ int i;
+
+ lines = 0;
+ for(i=0; i<m; i++) {
+ cb[k*lines++] = source[i];
+ }
+}
+
+/*---------------------------------------------------------------------------*\
+
+ quantise_init
+
+ Loads the entire LSP quantiser comprised of several vector quantisers
+ (codebooks).
+
+\*---------------------------------------------------------------------------*/
+
+void quantise_init()
+{
+ int i,k,m;
+
+ i = 0;
+ while(lsp_q[i].k) {
+ k = lsp_q[i].k;
+ m = lsp_q[i].m;
+ plsp_cb[i] = (float*)malloc(sizeof(float)*k*m);
+ assert(plsp_cb[i] != NULL);
+ load_cb(lsp_q[i].fn, plsp_cb[i], k, m);
+ i++;
+ assert(i < MAX_CB);
+ }
+}
+
+/*---------------------------------------------------------------------------*\
+
+ quantise
+
+ Quantises vec by choosing the nearest vector in codebook cb, and
+ returns the vector index. The squared error of the quantised vector
+ is added to se.
+
+\*---------------------------------------------------------------------------*/
+
+long quantise(float cb[], float vec[], float w[], int k, int m, float *se)
+/* float cb[][K]; current VQ codebook */
+/* float vec[]; vector to quantise */
+/* float w[]; weighting vector */
+/* int k; dimension of vectors */
+/* int m; size of codebook */
+/* float *se; accumulated squared error */
+{
+ float e; /* current error */
+ long besti; /* best index so far */
+ float beste; /* best error so far */
+ long j;
+ int i;
+
+ besti = 0;
+ beste = 1E32;
+ for(j=0; j<m; j++) {
+ e = 0.0;
+ for(i=0; i<k; i++)
+ e += pow((cb[j*k+i]-vec[i])*w[i],2.0);
+ if (e < beste) {
+ beste = e;
+ besti = j;
+ }
+ }
+
+ *se += beste;
+
+ return(besti);
+}
+
+static float gmin=PI;
+
+float get_gmin(void) { return gmin; }
+
+void min_lsp_dist(float lsp[], int order)
+{
+ int i;
+
+ for(i=1; i<order; i++)
+ if ((lsp[i]-lsp[i-1]) < gmin)
+ gmin = lsp[i]-lsp[i-1];
+}
+
+void check_lsp_order(float lsp[], int lpc_order)
+{
+ int i;
+ float tmp;
+
+ for(i=1; i<lpc_order; i++)
+ if (lsp[i] < lsp[i-1]) {
+ printf("swap %d\n",i);
+ tmp = lsp[i-1];
+ lsp[i-1] = lsp[i]-0.05;
+ lsp[i] = tmp+0.05;
+ }
+}
+
+void force_min_lsp_dist(float lsp[], int lpc_order)
+{
+ int i;
+
+ for(i=1; i<lpc_order; i++)
+ if ((lsp[i]-lsp[i-1]) < 0.01) {
+ lsp[i] += 0.01;
+ }
+}
+
+/*---------------------------------------------------------------------------*\
+
+ lpc_model_amplitudes
+
+ Derive a LPC model for amplitude samples then estimate amplitude samples
+ from this model with optional LSP quantisation.
+
+ Returns the spectral distortion for this frame.
+
+\*---------------------------------------------------------------------------*/
+
+float lpc_model_amplitudes(
+ float Sn[], /* Input frame of speech samples */
+ float w[],
+ MODEL *model, /* sinusoidal model parameters */
+ int order, /* LPC model order */
+ int lsp_quant, /* optional LSP quantisation if non-zero */
+ float ak[] /* output aks */
+)
+{
+ float Wn[M];
+ float R[LPC_MAX+1];
+ float E;
+ int i,j;
+ float snr;
+ float lsp[LPC_MAX];
+ float lsp_hz[LPC_MAX];
+ float lsp_[LPC_MAX];
+ int roots; /* number of LSP roots found */
+ int index;
+ float se;
+ int k,m;
+ float *cb;
+ float wt[LPC_MAX];
+
+ for(i=0; i<M; i++)
+ Wn[i] = Sn[i]*w[i];
+ autocorrelate(Wn,R,M,order);
+ levinson_durbin(R,ak,order);
+
+ E = 0.0;
+ for(i=0; i<=order; i++)
+ E += ak[i]*R[i];
+
+ for(i=0; i<order; i++)
+ wt[i] = 1.0;
+
+ if (lsp_quant) {
+ roots = lpc_to_lsp(ak, order, lsp, 5, LSP_DELTA1);
+ if (roots != order)
+ printf("LSP roots not found\n");
+
+ /* convert from radians to Hz to make quantisers more
+ human readable */
+
+ for(i=0; i<order; i++)
+ lsp_hz[i] = (4000.0/PI)*lsp[i];
+
+ /* simple uniform scalar quantisers */
+
+ for(i=0; i<10; i++) {
+ k = lsp_q[i].k;
+ m = lsp_q[i].m;
+ cb = plsp_cb[i];
+ index = quantise(cb, &lsp_hz[i], wt, k, m, &se);
+ lsp_hz[i] = cb[index*k];
+ }
+
+ /* experiment: simulating uniform quantisation error
+ for(i=0; i<order; i++)
+ lsp[i] += PI*(12.5/4000.0)*(1.0 - 2.0*(float)rand()/RAND_MAX);
+ */
+
+ for(i=0; i<order; i++)
+ lsp[i] = (PI/4000.0)*lsp_hz[i];
+
+ /* Bandwidth Expansion (BW). Prevents any two LSPs getting too
+ close together after quantisation. We know from experiment
+ that LSP quantisation errors < 12.5Hz (25Hz setp size) are
+ inaudible so we use that as the minimum LSP separation.
+ */
+
+ for(i=1; i<5; i++) {
+ if (lsp[i] - lsp[i-1] < PI*(12.5/4000.0))
+ lsp[i] = lsp[i-1] + PI*(12.5/4000.0);
+ }
+
+ /* as quantiser gaps increased, larger BW expansion was required
+ to prevent twinkly noises */
+
+ for(i=5; i<8; i++) {
+ if (lsp[i] - lsp[i-1] < PI*(25.0/4000.0))
+ lsp[i] = lsp[i-1] + PI*(25.0/4000.0);
+ }
+ for(i=8; i<order; i++) {
+ if (lsp[i] - lsp[i-1] < PI*(75.0/4000.0))
+ lsp[i] = lsp[i-1] + PI*(75.0/4000.0);
+ }
+
+ for(j=0; j<order; j++)
+ lsp_[j] = lsp[j];
+
+ lsp_to_lpc(lsp_, ak, order);
+ dump_lsp(lsp);
+ }
+
+ dump_E(E);
+ #ifdef SIM_QUANT
+ /* simulated LPC energy quantisation */
+ {
+ float e = 10.0*log10(E);
+ e += 2.0*(1.0 - 2.0*(float)rand()/RAND_MAX);
+ E = pow(10.0,e/10.0);
+ }
+ #endif
+
+ aks_to_M2(ak,order,model,E,&snr, 1); /* {ak} -> {Am} LPC decode */
+
+ return snr;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ aks_to_M2()
+
+ Transforms the linear prediction coefficients to spectral amplitude
+ samples. This function determines A(m) from the average energy per
+ band using an FFT.
+
+\*---------------------------------------------------------------------------*/
+
+void aks_to_M2(
+ float ak[], /* LPC's */
+ int order,
+ MODEL *model, /* sinusoidal model parameters for this frame */
+ float E, /* energy term */
+ float *snr, /* signal to noise ratio for this frame in dB */
+ int dump /* true to dump sample to dump file */
+)
+{
+ 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 */
+ float signal, noise;
+
+ 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 = ak[i];
+ four1(&Pw[-1].imag,FFT_DEC,1);
+
+ /* Determine power spectrum P(w) = E/(A(exp(jw))^2 ------------------------*/
+
+ for(i=0; i<FFT_DEC/2; i++)
+ Pw[i].real = E/(Pw[i].real*Pw[i].real + Pw[i].imag*Pw[i].imag);
+ if (dump)
+ dump_Pw(Pw);
+
+ /* Determine magnitudes by linear interpolation of P(w) -------------------*/
+
+ signal = noise = 0.0;
+ 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);
+ Em = 0.0;
+
+ for(i=am; i<bm; i++)
+ Em += Pw[i].real;
+ Am = sqrt(Em);
+
+ signal += pow(model->A[m],2.0);
+ noise += pow(model->A[m] - Am,2.0);
+ model->A[m] = Am;
+ }
+ *snr = 10.0*log10(signal/noise);
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: encode_Wo()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ Encodes Wo using a WO_LEVELS quantiser.
+
+\*---------------------------------------------------------------------------*/
+
+int encode_Wo(float Wo)
+{
+ int index;
+ float Wo_min = TWO_PI/P_MAX;
+ float Wo_max = TWO_PI/P_MIN;
+ float norm;
+
+ norm = (Wo - Wo_min)/(Wo_max - Wo_min);
+ index = floor(WO_LEVELS * norm + 0.5);
+ if (index < 0 ) index = 0;
+ if (index > (WO_LEVELS-1)) index = WO_LEVELS-1;
+
+ return index;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: decode_Wo()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ Decodes Wo using a WO_LEVELS quantiser.
+
+\*---------------------------------------------------------------------------*/
+
+float decode_Wo(int index)
+{
+ float Wo_min = TWO_PI/P_MAX;
+ float Wo_max = TWO_PI/P_MIN;
+ float step;
+ float Wo;
+
+ step = (Wo_max - Wo_min)/WO_LEVELS;
+ Wo = Wo_min + step*(index);
+
+ return Wo;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: speech_to_uq_lsps()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ Analyse a windowed frame of time domain speech to determine LPCs
+ which are the converted to LSPs for quantisation and transmission
+ over the channel.
+
+\*---------------------------------------------------------------------------*/
+
+float speech_to_uq_lsps(float lsp[],
+ float ak[],
+ float Sn[],
+ float w[],
+ int order
+)
+{
+ int i, roots;
+ float Wn[M];
+ float R[LPC_MAX+1];
+ float E;
+
+ for(i=0; i<M; i++)
+ Wn[i] = Sn[i]*w[i];
+ autocorrelate(Wn, R, M, order);
+ levinson_durbin(R, ak, order);
+
+ E = 0.0;
+ for(i=0; i<=order; i++)
+ E += ak[i]*R[i];
+
+ roots = lpc_to_lsp(ak, order, lsp, 5, LSP_DELTA1);
+ // assert(roots == order);
+
+ return E;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: encode_lsps()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ From a vector of unquantised (floating point) LSPs finds the quantised
+ LSP indexes.
+
+\*---------------------------------------------------------------------------*/
+
+void encode_lsps(int indexes[], float lsp[], int order)
+{
+ int i,k,m;
+ float wt[1];
+ float lsp_hz[LPC_MAX];
+ float *cb, se;
+
+ /* convert from radians to Hz so we can use human readable
+ frequencies */
+
+ for(i=0; i<order; i++)
+ lsp_hz[i] = (4000.0/PI)*lsp[i];
+
+ /* simple uniform scalar quantisers */
+
+ wt[0] = 1.0;
+ for(i=0; i<order; i++) {
+ k = lsp_q[i].k;
+ m = lsp_q[i].m;
+ cb = plsp_cb[i];
+ indexes[i] = quantise(cb, &lsp_hz[i], wt, k, m, &se);
+ }
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: decode_lsps()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ From a vector of quantised LSP indexes, returns the quantised
+ (floating point) LSPs.
+
+\*---------------------------------------------------------------------------*/
+
+void decode_lsps(float lsp[], int indexes[], int order)
+{
+ int i,k;
+ float lsp_hz[LPC_MAX];
+ float *cb;
+
+ for(i=0; i<order; i++) {
+ k = lsp_q[i].k;
+ cb = plsp_cb[i];
+ lsp_hz[i] = cb[indexes[i]*k];
+ }
+
+ /* convert back to radians */
+
+ for(i=0; i<order; i++)
+ lsp[i] = (PI/4000.0)*lsp_hz[i];
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: bw_expand_lsps()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ Applies Bandwidth Expansion (BW) to a vector of LSPs. Prevents any
+ two LSPs getting too close together after quantisation. We know
+ from experiment that LSP quantisation errors < 12.5Hz (25Hz setp
+ size) are inaudible so we use that as the minimum LSP separation.
+
+\*---------------------------------------------------------------------------*/
+
+void bw_expand_lsps(float lsp[],
+ int order
+)
+{
+ int i;
+
+ for(i=1; i<5; i++) {
+ if (lsp[i] - lsp[i-1] < PI*(12.5/4000.0))
+ lsp[i] = lsp[i-1] + PI*(12.5/4000.0);
+ }
+
+ /* As quantiser gaps increased, larger BW expansion was required
+ to prevent twinkly noises. This may need more experiment for
+ different quanstisers.
+ */
+
+ for(i=5; i<8; i++) {
+ if (lsp[i] - lsp[i-1] < PI*(25.0/4000.0))
+ lsp[i] = lsp[i-1] + PI*(25.0/4000.0);
+ }
+ for(i=8; i<order; i++) {
+ if (lsp[i] - lsp[i-1] < PI*(75.0/4000.0))
+ lsp[i] = lsp[i-1] + PI*(75.0/4000.0);
+ }
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: need_lpc_correction()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ Determine if we need LPC correction of first harmonic.
+
+\*---------------------------------------------------------------------------*/
+
+int need_lpc_correction(MODEL *model, float ak[], float E)
+{
+ MODEL tmp;
+ float snr,E1;
+
+ /* Find amplitudes so we can check if we need LPC correction.
+ TODO: replace call to aks_to_M2() by a single DFT calculation
+ of E/A(exp(jWo)) to make much more efficient. We only need
+ A[1].
+ */
+
+ memcpy(&tmp, model, sizeof(MODEL));
+ aks_to_M2(ak, LPC_ORD, &tmp, E, &snr, 0);
+
+ /*
+ Attenuate fundamental by 30dB if F0 < 150 Hz and LPC modelling
+ error for A[1] is larger than 6dB.
+
+ LPC modelling often makes big errors on 1st harmonic, for example
+ when the fundamental has been removed by analog high pass
+ filtering before sampling. However on unfiltered speech from
+ high quality sources we would like to keep the fundamental to
+ maintain the speech quality. So we check the error in A[1] and
+ attenuate it if the error is large to avoid annoying low
+ frequency energy after LPC modelling.
+
+ This requires a single bit to quantise, on top of the other
+ spectral magnitude bits (i.e. LSP bits + 1 total).
+ */
+
+ E1 = fabs(20.0*log10(model->A[1]) - 20.0*log10(tmp.A[1]));
+ if (E1 > 6.0)
+ return 1;
+ else
+ return 0;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: apply_lpc_correction()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ Apply first harmonic LPC correction at decoder.
+
+\*---------------------------------------------------------------------------*/
+
+void apply_lpc_correction(MODEL *model, int lpc_correction)
+{
+ if (lpc_correction) {
+ if (model->Wo < (PI*150.0/4000)) {
+ model->A[1] *= 0.032;
+ }
+ }
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: encode_energy()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ Encodes LPC energy using an E_LEVELS quantiser.
+
+\*---------------------------------------------------------------------------*/
+
+int encode_energy(float e)
+{
+ int index;
+ float e_min = E_MIN_DB;
+ float e_max = E_MAX_DB;
+ float norm;
+
+ e = 10.0*log10(e);
+ norm = (e - e_min)/(e_max - e_min);
+ index = floor(E_LEVELS * norm + 0.5);
+ if (index < 0 ) index = 0;
+ if (index > (E_LEVELS-1)) index = E_LEVELS-1;
+
+ return index;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: decode_energy()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ Decodes energy using a WO_BITS quantiser.
+
+\*---------------------------------------------------------------------------*/
+
+float decode_energy(int index)
+{
+ float e_min = E_MIN_DB;
+ float e_max = E_MAX_DB;
+ float step;
+ float e;
+
+ step = (e_max - e_min)/E_LEVELS;
+ e = e_min + step*(index);
+ e = pow(10.0,e/10.0);
+
+ return e;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: encode_amplitudes()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ Time domain LPC is used model the amplitudes which are then
+ converted to LSPs and quantised. So we don't actually encode the
+ amplitudes directly, rather we derive an equivalent representation
+ from the time domain speech.
+
+\*---------------------------------------------------------------------------*/
+
+void encode_amplitudes(int lsp_indexes[],
+ int *lpc_correction,
+ int *energy_index,
+ MODEL *model,
+ float Sn[],
+ float w[])
+{
+ float lsps[LPC_ORD];
+ float ak[LPC_ORD+1];
+ float e;
+
+ e = speech_to_uq_lsps(lsps, ak, Sn, w, LPC_ORD);
+ encode_lsps(lsp_indexes, lsps, LPC_ORD);
+ *lpc_correction = need_lpc_correction(model, ak, e);
+ *energy_index = encode_energy(e);
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: decode_amplitudes()
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/2010
+
+ Given the amplitude quantiser indexes recovers the harmonic
+ amplitudes.
+
+\*---------------------------------------------------------------------------*/
+
+float decode_amplitudes(MODEL *model,
+ float ak[],
+ int lsp_indexes[],
+ int lpc_correction,
+ int energy_index
+)
+{
+ float lsps[LPC_ORD];
+ float e;
+ float snr;
+
+ decode_lsps(lsps, lsp_indexes, LPC_ORD);
+ bw_expand_lsps(lsps, LPC_ORD);
+ lsp_to_lpc(lsps, ak, LPC_ORD);
+ e = decode_energy(energy_index);
+ aks_to_M2(ak, LPC_ORD, model, e, &snr, 1);
+ apply_lpc_correction(model, lpc_correction);
+
+ return snr;
+}
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: quantise.h
+ AUTHOR......: David Rowe
+ DATE CREATED: 31/5/92
+
+ Quantisation functions for the sinusoidal coder.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+ 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 __QUANTISE__
+#define __QUANTISE__
+
+#define WO_BITS 7
+#define WO_LEVELS (1<<WO_BITS)
+#define E_BITS 5
+#define E_LEVELS (1<<E_BITS)
+#define E_MIN_DB -10.0
+#define E_MAX_DB 40.0
+
+void quantise_init();
+float lpc_model_amplitudes(float Sn[], float w[], MODEL *model, int order,
+ int lsp,float ak[]);
+void aks_to_M2(float ak[], int order, MODEL *model, float E, float *snr,
+ int dump);
+float get_gmin(void);
+
+int encode_Wo(float Wo);
+float decode_Wo(int index);
+
+void encode_lsps(int indexes[], float lsp[], int order);
+void decode_lsps(float lsp[], int indexes[], int order);
+
+int encode_energy(float e);
+float decode_energy(int index);
+
+void encode_amplitudes(int lsp_indexes[],
+ int *lpc_correction,
+ int *energy_index,
+ MODEL *model,
+ float Sn[],
+ float w[]);
+
+float decode_amplitudes(MODEL *model,
+ float ak[],
+ int lsp_indexes[],
+ int lpc_correction,
+ int energy_index);
+
+void pack(unsigned char * bits, unsigned int *nbit, int index, unsigned int index_bits);
+int unpack(const unsigned char * bits, unsigned int *nbit, unsigned int index_bits);
+
+int lsp_bits(int i);
+
+int need_lpc_correction(MODEL *model, float ak[], float E);
+void apply_lpc_correction(MODEL *model, int lpc_correction);
+float speech_to_uq_lsps(float lsp[],
+ float ak[],
+ float Sn[],
+ float w[],
+ int order
+ );
+void bw_expand_lsps(float lsp[],
+ int order
+ );
+void decode_lsps(float lsp[], int indexes[], int order);
+
+#endif
--- /dev/null
+#!/bin/sh
+# sim.sh
+# David Rowe 10 Sep 2009
+
+# Process a source file using the codec 2 simulation. An output
+# speech file is generated for each major processing step, from the
+# unquantised siusoidal model to fully quantised. This way we can
+# listen to the effect of each processing step. Use listensim.sh to
+# test the output files.
+
+../src/c2sim ../raw/$1.raw -o $1_uq.raw
+../src/c2sim ../raw/$1.raw --phase0 -o $1_phase0.raw --postfilter
+../src/c2sim ../raw/$1.raw --lpc 10 -o $1_lpc10.raw
+../src/c2sim ../raw/$1.raw --lpc 10 --lsp -o $1_lsp.raw
+../src/c2sim ../raw/$1.raw --phase0 --lpc 10 -o $1_phase0_lpc10.raw --postfilter
+../src/c2sim ../raw/$1.raw --phase0 --lpc 10 --lsp -o $1_phase0_lsp.raw --postfilter
+../src/c2sim ../raw/$1.raw --phase0 --lpc 10 --lsp -o $1_phase0_lsp_dec.raw --postfilter --dec
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: sine.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 19/8/2010
+
+ Sinusoidal analysis and synthesis functions.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+ Copyright (C) 1990-2010 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.
+*/
+
+/*---------------------------------------------------------------------------*\
+
+ INCLUDES
+
+\*---------------------------------------------------------------------------*/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <math.h>
+
+#include "defines.h"
+#include "sine.h"
+#include "four1.h"
+
+/*---------------------------------------------------------------------------*\
+
+ HEADERS
+
+\*---------------------------------------------------------------------------*/
+
+void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax,
+ float pstep);
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTIONS
+
+\*---------------------------------------------------------------------------*/
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: make_analysis_window
+ AUTHOR......: David Rowe
+ DATE CREATED: 11/5/94
+
+ Init function that generates the time domain analysis window and it's DFT.
+
+\*---------------------------------------------------------------------------*/
+
+void make_analysis_window(float w[],COMP W[])
+{
+ float m;
+ COMP temp;
+ int i,j;
+
+ /*
+ Generate Hamming window centered on M-sample pitch analysis window
+
+ 0 M/2 M-1
+ |-------------|-------------|
+ |-------|-------|
+ NW samples
+
+ All our analysis/synthsis is centred on the M/2 sample.
+ */
+
+ m = 0.0;
+ for(i=0; i<M/2-NW/2; i++)
+ w[i] = 0.0;
+ for(i=M/2-NW/2,j=0; i<M/2+NW/2; i++,j++) {
+ w[i] = 0.5 - 0.5*cos(TWO_PI*j/(NW-1));
+ m += w[i]*w[i];
+ }
+ for(i=M/2+NW/2; i<M; i++)
+ w[i] = 0.0;
+
+ /* Normalise - makes freq domain amplitude estimation straight
+ forward */
+
+ m = 1.0/sqrt(m*FFT_ENC);
+ for(i=0; i<M; i++) {
+ w[i] *= m;
+ }
+
+ /*
+ Generate DFT of analysis window, used for later processing. Note
+ we modulo FFT_ENC shift the time domain window w[], this makes the
+ imaginary part of the DFT W[] equal to zero as the shifted w[] is
+ even about the n=0 time axis if NW is odd. Having the imag part
+ of the DFT W[] makes computation easier.
+
+ 0 FFT_ENC-1
+ |-------------------------|
+
+ ----\ /----
+ \ /
+ \ / <- shifted version of window w[n]
+ \ /
+ \ /
+ -------
+
+ |---------| |---------|
+ NW/2 NW/2
+ */
+
+ for(i=0; i<FFT_ENC; i++) {
+ W[i].real = 0.0;
+ W[i].imag = 0.0;
+ }
+ for(i=0; i<NW/2; i++)
+ W[i].real = w[i+M/2];
+ for(i=FFT_ENC-NW/2,j=M/2-NW/2; i<FFT_ENC; i++,j++)
+ W[i].real = w[j];
+
+ four1(&W[-1].imag,FFT_ENC,-1); /* "Numerical Recipes in C" FFT */
+
+ /*
+ Re-arrange W[] to be symmetrical about FFT_ENC/2. Makes later
+ analysis convenient.
+
+ Before:
+
+
+ 0 FFT_ENC-1
+ |----------|---------|
+ __ _
+ \ /
+ \_______________/
+
+ After:
+
+ 0 FFT_ENC-1
+ |----------|---------|
+ ___
+ / \
+ ________/ \_______
+
+ */
+
+
+ for(i=0; i<FFT_ENC/2; i++) {
+ temp.real = W[i].real;
+ temp.imag = W[i].imag;
+ W[i].real = W[i+FFT_ENC/2].real;
+ W[i].imag = W[i+FFT_ENC/2].imag;
+ W[i+FFT_ENC/2].real = temp.real;
+ W[i+FFT_ENC/2].imag = temp.imag;
+ }
+
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: dft_speech
+ AUTHOR......: David Rowe
+ DATE CREATED: 27/5/94
+
+ Finds the DFT of the current speech input speech frame.
+
+\*---------------------------------------------------------------------------*/
+
+void dft_speech(COMP Sw[], float Sn[], float w[])
+{
+ int i;
+
+ for(i=0; i<FFT_ENC; i++) {
+ Sw[i].real = 0.0;
+ Sw[i].imag = 0.0;
+ }
+
+ /* Centre analysis window on time axis, we need to arrange input
+ to FFT this way to make FFT phases correct */
+
+ /* move 2nd half to start of FFT input vector */
+
+ for(i=0; i<NW/2; i++)
+ Sw[i].real = Sn[i+M/2]*w[i+M/2];
+
+ /* move 1st half to end of FFT input vector */
+
+ for(i=0; i<NW/2; i++)
+ Sw[FFT_ENC-NW/2+i].real = Sn[i+M/2-NW/2]*w[i+M/2-NW/2];
+
+ four1(&Sw[-1].imag,FFT_ENC,-1);
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: two_stage_pitch_refinement
+ AUTHOR......: David Rowe
+ DATE CREATED: 27/5/94
+
+ Refines the current pitch estimate using the harmonic sum pitch
+ estimation technique.
+
+\*---------------------------------------------------------------------------*/
+
+void two_stage_pitch_refinement(MODEL *model, COMP Sw[])
+{
+ float pmin,pmax,pstep; /* pitch refinment minimum, maximum and step */
+
+ /* Coarse refinement */
+
+ pmax = TWO_PI/model->Wo + 5;
+ pmin = TWO_PI/model->Wo - 5;
+ pstep = 1.0;
+ hs_pitch_refinement(model,Sw,pmin,pmax,pstep);
+
+ /* Fine refinement */
+
+ pmax = TWO_PI/model->Wo + 1;
+ pmin = TWO_PI/model->Wo - 1;
+ pstep = 0.25;
+ hs_pitch_refinement(model,Sw,pmin,pmax,pstep);
+
+ /* Limit range */
+
+ if (model->Wo < TWO_PI/P_MAX)
+ model->Wo = TWO_PI/P_MAX;
+ if (model->Wo > TWO_PI/P_MIN)
+ model->Wo = TWO_PI/P_MIN;
+
+ model->L = floor(PI/model->Wo);
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: hs_pitch_refinement
+ AUTHOR......: David Rowe
+ DATE CREATED: 27/5/94
+
+ Harmonic sum pitch refinement function.
+
+ pmin pitch search range minimum
+ pmax pitch search range maximum
+ step pitch search step size
+ model current pitch estimate in model.Wo
+
+ model refined pitch estimate in model.Wo
+
+\*---------------------------------------------------------------------------*/
+
+void hs_pitch_refinement(MODEL *model, COMP Sw[], float pmin, float pmax, float pstep)
+{
+ int m; /* loop variable */
+ int b; /* bin for current harmonic centre */
+ float E; /* energy for current pitch*/
+ float Wo; /* current "test" fundamental freq. */
+ float Wom; /* Wo that maximises E */
+ float Em; /* mamimum energy */
+ float r; /* number of rads/bin */
+ float p; /* current pitch */
+
+ /* Initialisation */
+
+ model->L = PI/model->Wo; /* use initial pitch est. for L */
+ Wom = model->Wo;
+ Em = 0.0;
+ r = TWO_PI/FFT_ENC;
+
+ /* Determine harmonic sum for a range of Wo values */
+
+ for(p=pmin; p<=pmax; p+=pstep) {
+ E = 0.0;
+ Wo = TWO_PI/p;
+
+ /* Sum harmonic magnitudes */
+
+ for(m=1; m<=model->L; m++) {
+ b = floor(m*Wo/r + 0.5);
+ E += Sw[b].real*Sw[b].real + Sw[b].imag*Sw[b].imag;
+ }
+
+ /* Compare to see if this is a maximum */
+
+ if (E > Em) {
+ Em = E;
+ Wom = Wo;
+ }
+ }
+
+ model->Wo = Wom;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: estimate_amplitudes
+ AUTHOR......: David Rowe
+ DATE CREATED: 27/5/94
+
+ Estimates the complex amplitudes of the harmonics.
+
+\*---------------------------------------------------------------------------*/
+
+void estimate_amplitudes(MODEL *model, COMP Sw[], COMP W[])
+{
+ int i,m; /* loop variables */
+ int am,bm; /* bounds of current harmonic */
+ int b; /* DFT bin of centre of current harmonic */
+ float den; /* denominator of amplitude expression */
+ float r; /* number of rads/bin */
+ int offset;
+ COMP Am;
+
+ r = TWO_PI/FFT_ENC;
+
+ for(m=1; m<=model->L; m++) {
+ den = 0.0;
+ 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);
+
+ /* Estimate ampltude of harmonic */
+
+ den = 0.0;
+ Am.real = Am.imag = 0.0;
+ for(i=am; i<bm; i++) {
+ den += Sw[i].real*Sw[i].real + Sw[i].imag*Sw[i].imag;
+ offset = i + FFT_ENC/2 - floor(m*model->Wo/r + 0.5);
+ Am.real += Sw[i].real*W[offset].real;
+ Am.imag += Sw[i].imag*W[offset].real;
+ }
+
+ model->A[m] = sqrt(den);
+
+ /* Estimate phase of harmonic */
+
+ model->phi[m] = atan2(Sw[b].imag,Sw[b].real);
+ }
+}
+
+/*---------------------------------------------------------------------------*\
+
+ est_voicing_mbe()
+
+ Returns the error of the MBE cost function for a fiven F0.
+
+ Note: I think a lot of the operations below can be simplified as
+ W[].imag = 0 and has been normalised such that den always equals 1.
+
+\*---------------------------------------------------------------------------*/
+
+float est_voicing_mbe(
+ MODEL *model,
+ COMP Sw[],
+ COMP W[],
+ float f0,
+ COMP Sw_[] /* DFT of all voiced synthesised signal for f0 */
+ /* useful for debugging/dump file */
+)
+{
+ int i,l,al,bl,m; /* loop variables */
+ COMP Am; /* amplitude sample for this band */
+ int offset; /* centers Hw[] about current harmonic */
+ float den; /* denominator of Am expression */
+ float error; /* accumulated error between originl and synthesised */
+ float Wo; /* current "test" fundamental freq. */
+ int L;
+ float sig, snr;
+
+ sig = 0.0;
+ for(l=1; l<=model->L/4; l++) {
+ sig += model->A[l]*model->A[l];
+ }
+
+ for(i=0; i<FFT_ENC; i++) {
+ Sw_[i].real = 0.0;
+ Sw_[i].imag = 0.0;
+ }
+
+ L = floor((FS/2.0)/f0);
+ Wo = f0*(TWO_PI/FS);
+
+ error = 0.0;
+
+ /* Just test across the harmonics in the first 1000 Hz (L/4) */
+
+ for(l=1; l<=L/4; l++) {
+ Am.real = 0.0;
+ Am.imag = 0.0;
+ den = 0.0;
+ al = ceil((l - 0.5)*Wo*FFT_ENC/TWO_PI);
+ bl = ceil((l + 0.5)*Wo*FFT_ENC/TWO_PI);
+
+ /* Estimate amplitude of harmonic assuming harmonic is totally voiced */
+
+ for(m=al; m<bl; m++) {
+ offset = FFT_ENC/2 + m - l*Wo*FFT_ENC/TWO_PI + 0.5;
+ Am.real += Sw[m].real*W[offset].real + Sw[m].imag*W[offset].imag;
+ Am.imag += Sw[m].imag*W[offset].real - Sw[m].real*W[offset].imag;
+ den += W[offset].real*W[offset].real + W[offset].imag*W[offset].imag;
+ }
+
+ Am.real = Am.real/den;
+ Am.imag = Am.imag/den;
+
+ /* Determine error between estimated harmonic and original */
+
+ for(m=al; m<bl; m++) {
+ offset = FFT_ENC/2 + m - l*Wo*FFT_ENC/TWO_PI + 0.5;
+ Sw_[m].real = Am.real*W[offset].real - Am.imag*W[offset].imag;
+ Sw_[m].imag = Am.real*W[offset].imag + Am.imag*W[offset].real;
+ error += (Sw[m].real - Sw_[m].real)*(Sw[m].real - Sw_[m].real);
+ error += (Sw[m].imag - Sw_[m].imag)*(Sw[m].imag - Sw_[m].imag);
+ }
+ }
+
+ snr = 10.0*log10(sig/error);
+ if (snr > V_THRESH)
+ model->voiced = 1;
+ else
+ model->voiced = 0;
+
+ return snr;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: make_synthesis_window
+ AUTHOR......: David Rowe
+ DATE CREATED: 11/5/94
+
+ Init function that generates the trapezoidal (Parzen) sythesis window.
+
+\*---------------------------------------------------------------------------*/
+
+void make_synthesis_window(float Pn[])
+{
+ int i;
+ float win;
+
+ /* Generate Parzen window in time domain */
+
+ win = 0.0;
+ for(i=0; i<N/2-TW; i++)
+ Pn[i] = 0.0;
+ win = 0.0;
+ for(i=N/2-TW; i<N/2+TW; win+=1.0/(2*TW), i++ )
+ Pn[i] = win;
+ for(i=N/2+TW; i<3*N/2-TW; i++)
+ Pn[i] = 1.0;
+ win = 1.0;
+ for(i=3*N/2-TW; i<3*N/2+TW; win-=1.0/(2*TW), i++)
+ Pn[i] = win;
+ for(i=3*N/2+TW; i<2*N; i++)
+ Pn[i] = 0.0;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: synthesise
+ AUTHOR......: David Rowe
+ DATE CREATED: 20/2/95
+
+ Synthesise a speech signal in the frequency domain from the
+ sinusodal model parameters. Uses overlap-add a triangular window to
+ smoothly interpolate betwen frames.
+
+\*---------------------------------------------------------------------------*/
+
+void synthesise(
+ float Sn_[], /* time domain synthesised signal */
+ MODEL *model, /* ptr to model parameters for this frame */
+ float Pn[], /* time domain Parzen window */
+ int shift /* used to handle transition frames */
+)
+{
+ int i,l,j,b; /* loop variables */
+ COMP Sw_[FFT_DEC]; /* DFT of synthesised signal */
+
+ if (shift) {
+ /* Update memories */
+
+ for(i=0; i<N-1; i++) {
+ Sn_[i] = Sn_[i+N];
+ }
+ Sn_[N-1] = 0.0;
+ }
+
+ for(i=0; i<FFT_DEC; i++) {
+ Sw_[i].real = 0.0;
+ Sw_[i].imag = 0.0;
+ }
+
+ /* Now set up frequency domain synthesised speech */
+
+ for(l=1; l<=model->L; l++) {
+ b = floor(l*model->Wo*FFT_DEC/TWO_PI + 0.5);
+ Sw_[b].real = model->A[l]*cos(model->phi[l]);
+ Sw_[b].imag = model->A[l]*sin(model->phi[l]);
+ Sw_[FFT_DEC-b].real = Sw_[b].real;
+ Sw_[FFT_DEC-b].imag = -Sw_[b].imag;
+ }
+
+ /* Perform inverse DFT */
+
+ four1(&Sw_[-1].imag,FFT_DEC,1);
+
+ /* Overlap add to previous samples */
+
+ for(i=0; i<N-1; i++) {
+ Sn_[i] += Sw_[FFT_DEC-N+1+i].real*Pn[i];
+ }
+
+ if (shift)
+ for(i=N-1,j=0; i<2*N; i++,j++)
+ Sn_[i] = Sw_[j].real*Pn[i];
+ else
+ for(i=N-1,j=0; i<2*N; i++,j++)
+ Sn_[i] += Sw_[j].real*Pn[i];
+}
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: sine.h
+ AUTHOR......: David Rowe
+ DATE CREATED: 1/11/94
+
+ Header file for sinusoidal analysis and synthesis 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 __SINE__
+#define __SINE__
+
+void make_analysis_window(float w[], COMP W[]);
+void dft_speech(COMP Sw[], float Sn[], float w[]);
+void two_stage_pitch_refinement(MODEL *model, COMP Sw[]);
+void estimate_amplitudes(MODEL *model, COMP Sw[], COMP W[]);
+float est_voicing_mbe(MODEL *model, COMP Sw[], COMP W[], float f0, COMP Sw_[]);
+void make_synthesis_window(float Pn[]);
+void synthesise(float Sn_[], MODEL *model, float Pn[], int shift);
+
+#endif
--- /dev/null
+AM_CFLAGS = -I../src -g -DFLOATING_POINT -DVAR_ARRAYS
+AUTOMAKE_OPTS = gnu
+NAME = libcodec2
+AM_CPPFLAGS = $(AM_CFLAGS)
+
+bin_PROGRAMS = genres genlsp extract vqtrain tnlp tinterp tquant tcodec2
+
+
+genres_SOURCES = genres.c ../src/lpc.c ../src/codebook.c
+genres_LDADD = $(lib_LTLIBRARIES)
+genres_LDFLAGS = $(LIBS)
+
+genlsp_SOURCES = genlsp.c ../src/lpc.c ../src/lsp.c ../src/codebook.c
+genlsp_LDADD = $(lib_LTLIBRARIES)
+genlsp_LDFLAGS = $(LIBS)
+
+extract_SOURCES = extract.c ../src/codebook.c
+extract_LDADD = $(lib_LTLIBRARIES)
+extract_LDFLAGS = $(LIBS)
+
+vqtrain_SOURCES = vqtrain.c ../src/codebook.c
+vqtrain_LDADD = $(lib_LTLIBRARIES)
+vqtrain_LDFLAGS = $(LIBS)
+
+tnlp_SOURCES = tnlp.c ../src/sine.c ../src/nlp.c ../src/four1.c ../src/dump.c ../src/codebook.c
+tnlp_LDADD = $(lib_LTLIBRARIES)
+tnlp_LDFLAGS = $(LIBS)
+
+tinterp_SOURCES = tinterp.c ../src/sine.c ../src/four1.c ../src/interp.c ../src/codebook.c
+tinterp_LDADD = $(lib_LTLIBRARIES)
+tinterp_LDFLAGS = $(LIBS)
+
+tquant_SOURCES = tquant.c ../src/quantise.c ../src/lpc.c ../src/lsp.c ../src/dump.c ../src/four1.c ../src/codebook.c
+tquant_LDADD = $(lib_LTLIBRARIES)
+tquant_LDFLAGS = $(LIBS)
+
+tcodec2_SOURCES = tcodec2.c ../src/quantise.c ../src/lpc.c ../src/lsp.c ../src/dump.c ../src/four1.c \
+../src/codec2.c ../src/sine.c ../src/nlp.c ../src/postfilter.c ../src/phase.c ../src/interp.c ../src/pack.c ../src/codebook.c
+tcodec2_LDADD = $(lib_LTLIBRARIES)
+tcodec2_LDFLAGS = $(LIBS)
+
--- /dev/null
+# Makefile.in generated by automake 1.9.6 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994, 1995, 1996, 1997, 1998, 1999, 2000, 2001, 2002,
+# 2003, 2004, 2005 Free Software Foundation, Inc.
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
+@SET_MAKE@
+
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+POST_UNINSTALL = :
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+host_triplet = @host@
+bin_PROGRAMS = genres$(EXEEXT) genlsp$(EXEEXT) extract$(EXEEXT) \
+ vqtrain$(EXEEXT) tnlp$(EXEEXT) tinterp$(EXEEXT) \
+ tquant$(EXEEXT) tcodec2$(EXEEXT)
+subdir = unittest
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+mostlyclean-libtool:
+ -rm -f *.lo
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+mostlyclean-generic:
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+
+# 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:
--- /dev/null
+/*--------------------------------------------------------------------------*\
+
+ FILE........: extract.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/2/95
+
+ This program extracts a float file of vectors from a text file
+ of vectors. The float files are easier to process quickly
+ during VQ training. A subset of the text file VQ may be
+ extracted to faciltate split VQ of scaler VQ design.
+
+\*--------------------------------------------------------------------------*/
+
+/*
+ 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, 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 MAX_STR 256 /* maximum string length */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <ctype.h>
+
+void scan_line(FILE *fp, float f[], int n);
+
+int main(int argc, char *argv[]) {
+ FILE *ftext; /* text file of vectors */
+ FILE *ffloat; /* float file of vectors */
+ int st,en; /* start and end values of vector to copy */
+ float *buf; /* ptr to vector read from ftext */
+ long lines; /* lines read so far */
+
+ if (argc != 5) {
+ printf("usage: extract TextFile FloatFile start end\n");
+ exit(0);
+ }
+
+ /* read command line arguments and open files */
+
+ ftext = fopen(argv[1],"rt");
+ if (ftext == NULL) {
+ printf("Error opening text file: %s\n",argv[1]);
+ exit(1);
+ }
+
+ ffloat = fopen(argv[2],"wb");
+ if (ffloat == NULL) {
+ printf("Error opening float file: %s\n",argv[2]);
+ exit(1);
+ }
+
+ st = atoi(argv[3]);
+ en = atoi(argv[4]);
+
+ buf = (float*)malloc(en*sizeof(float));
+ if (buf == NULL) {
+ printf("Error in malloc()\n");
+ exit(1);
+ }
+
+ lines = 0;
+ while(!feof(ftext)) {
+ scan_line(ftext, buf, en);
+ fwrite(&buf[st-1], sizeof(float), en-st+1, ffloat);
+ printf("\r%ld lines",lines++);
+ }
+ printf("\n");
+
+ /* clean up and exit */
+
+ free(buf);
+ fclose(ftext);
+ fclose(ffloat);
+
+ return 0;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: scan_line()
+
+ AUTHOR......: David Rowe
+ DATE CREATED: 20/2/95
+
+ This function reads a vector of floats from a line in a text file.
+
+\*---------------------------------------------------------------------------*/
+
+void scan_line(FILE *fp, float f[], int n)
+/* FILE *fp; file ptr to text file */
+/* float f[]; array of floats to return */
+/* int n; number of floats in line */
+{
+ char s[MAX_STR];
+ char *ps,*pe;
+ int i;
+
+ fgets(s,MAX_STR,fp);
+ ps = pe = s;
+ for(i=0; i<n; i++) {
+ while( isspace(*pe)) pe++;
+ while( !isspace(*pe)) pe++;
+ sscanf(ps,"%f",&f[i]);
+ ps = pe;
+ }
+}
+
--- /dev/null
+/*--------------------------------------------------------------------------*\
+
+ FILE........: genlsp.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/2/95
+
+ This program genrates a text file of LSP vectors from an input
+ speech file.
+
+\*--------------------------------------------------------------------------*/
+
+/*
+ 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, 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 P 10 /* LP order */
+#define LSP_DELTA1 0.05 /* grid spacing for LSP root searches */
+#define NW 279 /* frame size in samples */
+#define N 80 /* frame to frame shift */
+#define THRESH 40.0 /* threshold energy/sample for frame inclusion */
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+#include "lpc.h" /* LPC analysis functions */
+#include "lsp.h" /* LSP encode/decode functions */
+
+int switch_present(sw,argc,argv)
+ char sw[]; /* switch in string form */
+ int argc; /* number of command line arguments */
+ char *argv[]; /* array of command line arguments in string form */
+{
+ int i; /* loop variable */
+
+ for(i=1; i<argc; i++)
+ if (!strcmp(sw,argv[i]))
+ return(i);
+
+ return 0;
+}
+
+int main(int argc, char *argv[]) {
+ FILE *fspc; /* input file ptr for test database */
+ FILE *flsp; /* output text file of LSPs */
+ short buf[N]; /* input frame of speech samples */
+ float Sn[NW]; /* float input speech samples */
+ float ak[P+1]; /* LPCs for current frame */
+ float lsp[P]; /* LSPs for current frame */
+ float E; /* frame energy */
+ long af; /* number frames with "active" speech */
+ float Eres; /* LPC residual energy */
+ int i;
+ int roots;
+ int unstables;
+ int lspd;
+
+ /* Initialise ------------------------------------------------------*/
+
+ if (argc < 3) {
+ printf("usage: gentest RawFile LSPTextFile [--lspd]\n");
+ exit(0);
+ }
+
+ /* Open files */
+
+ fspc = fopen(argv[1],"rb");
+ if (fspc == NULL) {
+ printf("Error opening input SPC file: %s",argv[1]);
+ exit(1);
+ }
+
+ flsp = fopen(argv[2],"wt");
+ if (flsp == NULL) {
+ printf("Error opening output LSP file: %s",argv[2]);
+ exit(1);
+ }
+
+ lspd = switch_present("--lspd", argc, argv);
+
+ for(i=0; i<NW; i++)
+ Sn[i] = 0.0;
+
+ /* Read SPC file, and determine aks[] for each frame ------------------*/
+
+ af = 0;
+ unstables = 0;
+ while(fread(buf,sizeof(short),N,fspc) == N) {
+
+ for(i=0; i<NW-N; i++)
+ Sn[i] = Sn[i+N];
+ E = 0.0;
+ for(i=0; i<N; i++) {
+ Sn[i+NW-N] = buf[i];
+ E += Sn[i]*Sn[i];
+ }
+
+ E = 0.0;
+ for(i=0; i<NW; i++) {
+ E += Sn[i]*Sn[i];
+ }
+ E = 10.0*log10(E/NW);
+
+ /* If energy high enough, include this frame */
+
+ if (E > THRESH) {
+ af++;
+ printf("Active Frame: %ld unstables: %d\n",af, unstables);
+
+ find_aks(Sn, ak, NW, P, &Eres);
+ roots = lpc_to_lsp(&ak[1], P , lsp, 5, LSP_DELTA1);
+ if (roots == P) {
+ if (lspd) {
+ fprintf(flsp,"%f ",lsp[0]);
+ for(i=1; i<P; i++)
+ fprintf(flsp,"%f ",lsp[i]-lsp[i-1]);
+ fprintf(flsp,"\n");
+ }
+ else {
+ for(i=0; i<P; i++)
+ fprintf(flsp,"%f ",lsp[i]);
+ fprintf(flsp,"\n");
+ }
+ }
+ else
+ unstables++;
+ }
+ }
+
+ fclose(fspc);
+ fclose(flsp);
+
+ return 0;
+}
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: genres.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 24/8/09
+
+ Generates a file of LPC residual samples from original speech.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+ 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, 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 <stdio.h>
+#include <stdlib.h>
+#include <lpc.h>
+
+#define N 160
+#define P 10
+
+int main(int argc, char *argv[])
+{
+ FILE *fin,*fres; /* input and output files */
+ short buf[N]; /* buffer of 16 bit speech samples */
+ float Sn[P+N]; /* input speech samples */
+ float res[N]; /* residual after LPC filtering */
+ float E;
+ float ak[P+1]; /* LP coeffs */
+
+ int frames; /* frames processed so far */
+ int i; /* loop variables */
+
+ if (argc < 3) {
+ printf("usage: %s InputFile ResidualFile\n", argv[0]);
+ exit(0);
+ }
+
+ /* Open files */
+
+ if ((fin = fopen(argv[1],"rb")) == NULL) {
+ printf("Error opening input file: %s\n",argv[1]);
+ exit(0);
+ }
+
+ if ((fres = fopen(argv[2],"wb")) == NULL) {
+ printf("Error opening output residual file: %s\n",argv[2]);
+ exit(0);
+ }
+
+ /* Initialise */
+
+ frames = 0;
+ for(i=0; i<P; i++) {
+ Sn[i] = 0.0;
+ }
+
+ /* Main loop */
+
+ while( (fread(buf,sizeof(short),N,fin)) == N) {
+ frames++;
+ for(i=0; i<N; i++)
+ Sn[P+i] = (float)buf[i];
+
+ /* Determine {ak} and filter to find residual */
+
+ find_aks(&Sn[P], ak, N, P, &E);
+ inverse_filter(&Sn[P], ak, N, res, P);
+ for(i=0; i<N; i++)
+ buf[i] = (short)res[i];
+ fwrite(buf,sizeof(short),N,fres);
+ }
+
+ fclose(fin);
+ fclose(fres);
+
+ return 0;
+}
--- /dev/null
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--- /dev/null
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--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: lsptest.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 24/8/09
+
+ Test Speech LPC to LSP conversion and quantisation.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+ 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, 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 <lpc.h>
+#include <lsp.h>
+#include <sd.h>
+
+#define N 160
+#define P 10
+
+#define LPC_FLOOR 0.0002 /* autocorrelation floor */
+#define LSP_DELTA1 0.2 /* grid spacing for LSP root searches */
+#define NDFT 256 /* DFT size for SD calculation */
+
+/* Speex lag window */
+
+const float lag_window[11] = {
+ 1.00000, 0.99716, 0.98869, 0.97474, 0.95554, 0.93140, 0.90273, 0.86998,
+ 0.83367, 0.79434, 0.75258
+};
+
+/*---------------------------------------------------------------------------*\
+
+ find_aks_for_lsp()
+
+ This function takes a frame of samples, and determines the linear
+ prediction coefficients for that frame of samples. Modified version of
+ find_aks from lpc.c to include autocorrelation noise floor and lag window
+ to match Speex processing steps prior to LSP conversion.
+
+\*---------------------------------------------------------------------------*/
+
+void find_aks_for_lsp(
+ 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[N]; /* windowed frame of Nsam speech samples */
+ float R[P+1]; /* order+1 autocorrelation values of Sn[] */
+ int i;
+
+ hanning_window(Sn,Wn,Nsam);
+
+ autocorrelate(Wn,R,Nsam,order);
+ R[0] += LPC_FLOOR;
+ assert(order == 10); /* lag window only defined for order == 10 */
+ for(i=0; i<=order; i++)
+ R[i] *= lag_window[i];
+ 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;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ MAIN
+
+\*---------------------------------------------------------------------------*/
+
+int main(int argc, char *argv[])
+{
+ FILE *fin; /* input speech files */
+ short buf[N]; /* buffer of 16 bit speech samples */
+ float Sn[P+N]; /* input speech samples */
+ float E;
+ float ak[P+1]; /* LP coeffs */
+ float ak_[P+1]; /* quantised LP coeffs */
+ float lsp[P];
+ float lsp_[P]; /* quantised LSPs */
+ int roots; /* number of LSP roots found */
+ int frames; /* frames processed so far */
+ int i; /* loop variables */
+
+ SpeexBits bits;
+
+ float sd; /* SD for this frame */
+ float totsd; /* accumulated SD so far */
+ int gt2,gt4; /* number of frames > 2 and 4 dB SD */
+ int unstables; /* number of unstable LSP frames */
+
+ if (argc < 2) {
+ printf("usage: %s InputFile\n", argv[0]);
+ exit(0);
+ }
+
+ /* Open files */
+
+ if ((fin = fopen(argv[1],"rb")) == NULL) {
+ printf("Error opening input file: %s\n",argv[1]);
+ exit(0);
+ }
+
+ /* Initialise */
+
+ frames = 0;
+ for(i=0; i<P; i++) {
+ Sn[i] = 0.0;
+ }
+ ak_[0] = 1.0;
+
+ speex_bits_init(&bits);
+
+ totsd = 0.0;
+ unstables = 0;
+ gt2 = 0; gt4 = 0;
+
+ /* Main loop */
+
+ while( (fread(buf,sizeof(short),N,fin)) == N) {
+ frames++;
+ for(i=0; i<N; i++)
+ Sn[P+i] = (float)buf[i];
+
+ /* convert to LSP domain and back */
+
+ find_aks(&Sn[P], ak, N, P, &E);
+ roots = lpc_to_lsp(&ak[1], P , lsp, 10, LSP_DELTA1, NULL);
+ if (roots == P) {
+
+ speex_bits_reset(&bits);
+ lsp_quant_lbr(lsp, lsp_, P, &bits);
+ lsp_to_lpc(lsp_, &ak_[1], P, NULL);
+
+ /* measure spectral distortion */
+ sd = spectral_dist(ak, ak_, P, NDFT);
+ if (sd > 2.0) gt2++;
+ if (sd > 4.0) gt4++;
+ totsd += sd;
+ }
+ else
+ unstables++;
+ }
+
+ fclose(fin);
+
+ printf("frames = %d Av sd = %3.2f dB", frames, totsd/frames);
+ printf(" >2 dB %3.2f%% >4 dB %3.2f%% unstables: %d\n",gt2*100.0/frames,
+ gt4*100.0/frames, unstables);
+
+ return 0;
+}
+
--- /dev/null
+/*--------------------------------------------------------------------------*\
+
+ FILE........: sd.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 20/7/93
+
+ Function to determine spectral distortion between two sets of LPCs.
+
+\*--------------------------------------------------------------------------*/
+
+/*
+ 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, 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 MAX_N 2048 /* maximum DFT size */
+
+#include <math.h>
+#include "four1.h"
+#include "comp.h"
+#include "sd.h"
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: spectral_dist()
+
+ AUTHOR......: David Rowe
+ DATE CREATED: 20/7/93
+
+ This function returns the soectral distoertion between two
+ sets of LPCs.
+
+\*---------------------------------------------------------------------------*/
+
+float spectral_dist(float ak1[], float ak2[], int p, int n)
+/* float ak1[]; unquantised set of p+1 LPCs */
+/* float ak2[]; quantised set of p+1 LPCs */
+/* int p; LP order */
+/* int n; DFT size to use for SD calculations (power of 2) */
+{
+ COMP A1[MAX_N]; /* DFT of ak1[] */
+ COMP A2[MAX_N]; /* DFT of ak2[] */
+ float P1,P2; /* power of current bin */
+ float sd;
+ int i;
+
+ for(i=0; i<n; i++) {
+ A1[i].real = 0.0;
+ A1[i].imag = 0.0;
+ A2[i].real = 0.0;
+ A2[i].imag = 0.0;
+ }
+
+ for(i=0; i<p+1; i++) {
+ A1[i].real = ak1[i];
+ A2[i].real = ak2[i];
+ }
+
+ four1(&A1[-1].imag,n,-1);
+ four1(&A2[-1].imag,n,-1);
+
+ sd = 0.0;
+ for(i=0; i<n; i++) {
+ P1 = A1[i].real*A1[i].real + A1[i].imag*A1[i].imag;
+ P2 = A2[i].real*A2[i].real + A2[i].imag*A2[i].imag;
+ sd += pow(log10(P2/P1),2.0);
+ }
+ sd = 10.0*sqrt(sd/n); /* sd in dB */
+
+ return(sd);
+}
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: tcodec2.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 24/8/10
+
+ Test program for codec2.c functions.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+ 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, 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 <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+#include "defines.h"
+#include "codec2.h"
+#include "quantise.h"
+#include "interp.h"
+
+/* CODEC2 struct copies from codec2.c to help with testing */
+
+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 */
+} CODEC2;
+
+void analyse_one_frame(CODEC2 *c2, MODEL *model, short speech[]);
+void synthesise_one_frame(CODEC2 *c2, short speech[], MODEL *model, float ak[]);
+
+int test1()
+{
+ FILE *fin, *fout;
+ short buf[N];
+ void *c2;
+ CODEC2 *c3;
+ MODEL model;
+ float ak[LPC_ORD+1];
+ float lsps[LPC_ORD];
+
+ c2 = codec2_create();
+ c3 = (CODEC2*)c2;
+
+ fin = fopen("../raw/hts1a.raw", "rb");
+ assert(fin != NULL);
+ fout = fopen("hts1a_test.raw", "wb");
+ assert(fout != NULL);
+
+ while(fread(buf, sizeof(short), N, fin) == N) {
+ analyse_one_frame(c3, &model, buf);
+ speech_to_uq_lsps(lsps, ak, c3->Sn, c3->w, LPC_ORD);
+ synthesise_one_frame(c3, buf, &model, ak);
+ fwrite(buf, sizeof(short), N, fout);
+ }
+
+ codec2_destroy(c2);
+
+ fclose(fin);
+ fclose(fout);
+
+ return 0;
+}
+
+int test2()
+{
+ FILE *fin, *fout;
+ short buf[2*N];
+ void *c2;
+ CODEC2 *c3;
+ MODEL model, model_interp;
+ float ak[LPC_ORD+1];
+ int voiced1, voiced2;
+ int lsp_indexes[LPC_ORD];
+ int lpc_correction;
+ int energy_index;
+ int Wo_index;
+ char bits[CODEC2_BITS_PER_FRAME];
+ int nbit;
+ int i;
+
+ c2 = codec2_create();
+ c3 = (CODEC2*)c2;
+
+ fin = fopen("../raw/hts1a.raw", "rb");
+ assert(fin != NULL);
+ fout = fopen("hts1a_test.raw", "wb");
+ assert(fout != NULL);
+
+ while(fread(buf, sizeof(short), 2*N, fin) == 2*N) {
+ /* first 10ms analysis frame - we just want voicing */
+
+ analyse_one_frame(c3, &model, buf);
+ voiced1 = model.voiced;
+
+ /* second 10ms analysis frame */
+
+ analyse_one_frame(c3, &model, &buf[N]);
+ voiced2 = model.voiced;
+
+ Wo_index = encode_Wo(model.Wo);
+ encode_amplitudes(lsp_indexes,
+ &lpc_correction,
+ &energy_index,
+ &model,
+ c3->Sn,
+ c3->w);
+ nbit = 0;
+ 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);
+
+ nbit = 0;
+ 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);
+
+ 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, &c3->prev_model, &model);
+
+ synthesise_one_frame(c3, buf, &model_interp, ak);
+ synthesise_one_frame(c3, &buf[N], &model, ak);
+
+ memcpy(&c3->prev_model, &model, sizeof(MODEL));
+ fwrite(buf, sizeof(short), 2*N, fout);
+ }
+
+ codec2_destroy(c2);
+
+ fclose(fin);
+ fclose(fout);
+
+ return 0;
+}
+
+int test3()
+{
+ FILE *fin, *fout, *fbits;
+ short buf1[2*N];
+ short buf2[2*N];
+ char bits[CODEC2_BITS_PER_FRAME];
+ void *c2;
+
+ c2 = codec2_create();
+
+ fin = fopen("../raw/hts1a.raw", "rb");
+ assert(fin != NULL);
+ fout = fopen("hts1a_test.raw", "wb");
+ assert(fout != NULL);
+ fbits = fopen("hts1a_test3.bit", "wb");
+ assert(fout != NULL);
+
+ while(fread(buf1, sizeof(short), 2*N, fin) == 2*N) {
+ codec2_encode(c2, bits, buf1);
+ fwrite(bits, sizeof(char), CODEC2_BITS_PER_FRAME, fbits);
+ codec2_decode(c2, buf2, bits);
+ fwrite(buf2, sizeof(short), CODEC2_SAMPLES_PER_FRAME, fout);
+ }
+
+ codec2_destroy(c2);
+
+ fclose(fin);
+ fclose(fout);
+ fclose(fbits);
+
+ return 0;
+}
+
+int main() {
+ test3();
+ return 0;
+}
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: tcontphase.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 11/9/09
+
+ Test program for developing continuous phase track synthesis algorithm.
+ However while developing this it was discovered that synthesis_mixed()
+ worked just as well.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+ 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, 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 N 80 /* frame size */
+#define F 160 /* frames to synthesis */
+#define P 10 /* LPC order */
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+#include "sine.h"
+#include "dump.h"
+#include "synth.h"
+#include "phase.h"
+
+int frames;
+
+float ak[] = {
+ 1.000000,
+-1.455836,
+ 1.361841,
+-0.879267,
+ 0.915985,
+-1.002202,
+ 0.944103,
+-0.743094,
+ 1.053356,
+-0.817491,
+ 0.431222
+};
+
+
+/*---------------------------------------------------------------------------*\
+
+ 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)
+ char sw[]; /* switch in string form */
+ int argc; /* number of command line arguments */
+ char *argv[]; /* array of command line arguments in string form */
+{
+ int i; /* loop variable */
+
+ for(i=1; i<argc; i++)
+ if (!strcmp(sw,argv[i]))
+ return(i);
+
+ return 0;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ MAIN
+
+\*---------------------------------------------------------------------------*/
+
+int main(argc,argv)
+int argc;
+char *argv[];
+{
+ FILE *fout;
+ short buf[N];
+ int i,j;
+ int dump;
+ float phi_prev[MAX_AMP];
+ float Wo_prev, ex_phase, G;
+ //float ak[P+1];
+ COMP H[MAX_AMP];
+ float f0;
+
+ if (argc < 3) {
+ printf("\nusage: tcontphase OutputRawSpeechFile F0\n");
+ exit(0);
+ }
+
+ /* Output file */
+
+ if ((fout = fopen(argv[1],"wb")) == NULL) {
+ printf("Error opening output speech file: %s\n",argv[1]);
+ exit(1);
+ }
+
+ f0 = atof(argv[2]);
+
+ dump = switch_present("--dump",argc,argv);
+ if (dump)
+ dump_on(argv[dump+1]);
+
+ init_decoder();
+
+ for(i=0; i<MAX_AMP; i++)
+ phi_prev[i] = 0.0;
+ Wo_prev = 0.0;
+
+ model.Wo = PI*(f0/4000.0);
+ G = 1000.0;
+ model.L = floor(PI/model.Wo);
+
+ //aks_to_H(&model, ak, G , H, P);
+ //for(i=1; i<=model.L; i++)
+ model.A[i] = sqrt(H[i].real*H[i].real + H[i].imag*H[i].imag);
+ //printf("L = %d\n", model.L);
+ //model.L = 10;
+ for(i=1; i<=model.L; i++) {
+ model.A[i] = 1000/model.L;
+ model.phi[i] = 0;
+ H[i].real = 1.0; H[i].imag = 0.0;
+ }
+
+ //ak[0] = 1.0;
+ //for(i=1; i<=P; i++)
+ // ak[i] = 0.0;
+
+ frames = 0;
+ for(j=0; j<F; j++) {
+ frames++;
+
+ #ifdef SWAP
+ /* lets make phases bounce around from frame to frame. This
+ could happen if H[m] is varying, for example due to frame
+ to frame Wo variations, or non-stationary speech.
+ Continous model generally results in smooth phase track
+ under these circumstances. */
+ if (j%2){
+ H[1].real = 1.0; H[1].imag = 0.0;
+ model.phi[1] = 0.0;
+ }
+ else {
+ H[1].real = 0.0; H[1].imag = 1.0;
+ model.phi[1] = PI/2;
+ }
+ #endif
+
+ //#define CONT
+ #ifdef CONT
+ synthesise_continuous_phase(Pn, &model, Sn_, 1, &Wo_prev, phi_prev);
+ #else
+ phase_synth_zero_order(5.0, H, &Wo_prev, &ex_phase);
+ synthesise_mixed(Pn,&model,Sn_,1);
+ #endif
+
+ for(i=0; i<N; i++)
+ buf[i] = Sn_[i];
+ fwrite(buf,sizeof(short),N,fout);
+ }
+
+ fclose(fout);
+ if (dump) dump_off();
+
+ return 0;
+}
+
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: tinterp.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/10
+
+ Tests interpolation functions.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+ 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, 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 <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+#include <ctype.h>
+
+#include "defines.h"
+#include "sine.h"
+#include "interp.h"
+
+void make_amp(MODEL *model, float f0, float cdB, float mdBHz)
+{
+ int i;
+ float mdBrad = mdBHz*FS/TWO_PI;
+
+ model->Wo = f0*TWO_PI/FS;
+ model->L = PI/model->Wo;
+ for(i=0; i<=model->L; i++)
+ model->A[i] = pow(10.0,(cdB + (float)i*model->Wo*mdBrad)/20.0);
+ model->voiced = 1;
+}
+
+void write_amp(char file[], MODEL *model)
+{
+ FILE *f;
+ int i;
+
+ f = fopen(file,"wt");
+ for(i=1; i<=model->L; i++)
+ fprintf(f, "%f\t%f\n", model->Wo*i, model->A[i]);
+ fclose(f);
+}
+
+char *get_next_float(char *s, float *num)
+{
+ char *p = s;
+ char tmp[MAX_STR];
+
+ while(*p && !isspace(*p))
+ p++;
+ memcpy(tmp, s, p-s);
+ tmp[p-s] = 0;
+ *num = atof(tmp);
+
+ return p+1;
+}
+
+char *get_next_int(char *s, int *num)
+{
+ char *p = s;
+ char tmp[MAX_STR];
+
+ while(*p && !isspace(*p))
+ p++;
+ memcpy(tmp, s, p-s);
+ tmp[p-s] = 0;
+ *num = atoi(tmp);
+
+ return p+1;
+}
+
+void load_amp(MODEL *model, char file[], int frame)
+{
+ FILE *f;
+ int i;
+ char s[1024];
+ char *ps;
+
+ f = fopen(file,"rt");
+
+ for(i=0; i<frame; i++)
+ fgets(s, 1023, f);
+
+ ps = s;
+ ps = get_next_float(ps, &model->Wo);
+ ps = get_next_int(ps, &model->L);
+ for(i=1; i<=model->L; i++)
+ ps = get_next_float(ps, &model->A[i]);
+
+ fclose(f);
+}
+
+int main() {
+ MODEL prev, next, interp;
+
+ //make_amp(&prev, 50.0, 60.0, 6E-3);
+ //make_amp(&next, 50.0, 40.0, 6E-3);
+ load_amp(&prev, "../src/hts1a_model.txt", 32);
+ load_amp(&next, "../src/hts1a_model.txt", 34);
+
+ interp.voiced = 1;
+ interpolate(&interp, &prev, &next);
+
+ write_amp("tinterp_prev.txt", &prev);
+ write_amp("tinterp_interp.txt", &interp);
+ write_amp("tinterp_next.txt", &next);
+
+ return 0;
+}
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: tnlp.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/3/93
+
+ Test program for non linear pitch 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, 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 N 80 /* frame size */
+#define M 320 /* pitch analysis window size */
+#define PITCH_MIN 20
+#define PITCH_MAX 160
+#define TNLP
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+
+#include "defines.h"
+#include "dump.h"
+#include "sine.h"
+#include "nlp.h"
+
+int frames;
+
+/*---------------------------------------------------------------------------*\
+
+ 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)
+ char sw[]; /* switch in string form */
+ int argc; /* number of command line arguments */
+ char *argv[]; /* array of command line arguments in string form */
+{
+ int i; /* loop variable */
+
+ for(i=1; i<argc; i++)
+ if (!strcmp(sw,argv[i]))
+ return(i);
+
+ return 0;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ MAIN
+
+\*---------------------------------------------------------------------------*/
+
+int main(argc,argv)
+int argc;
+char *argv[];
+{
+ FILE *fin,*fout;
+ short buf[N];
+ 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[] */
+ float pitch;
+ int i;
+ int dump;
+ float prev_Wo;
+ void *nlp_states;
+
+ if (argc < 3) {
+ printf("\nusage: tnlp InputRawSpeechFile OutputPitchTextFile "
+ "[--dump DumpFile]\n");
+ exit(0);
+ }
+
+ /* Input file */
+
+ if ((fin = fopen(argv[1],"rb")) == NULL) {
+ printf("Error opening input speech file: %s\n",argv[1]);
+ exit(1);
+ }
+
+ /* Output file */
+
+ if ((fout = fopen(argv[2],"wt")) == NULL) {
+ printf("Error opening output text file: %s\n",argv[2]);
+ exit(1);
+ }
+
+ dump = switch_present("--dump",argc,argv);
+ if (dump)
+ dump_on(argv[dump+1]);
+
+ nlp_states = nlp_create();
+ make_analysis_window(w,W);
+
+ frames = 0;
+ prev_Wo = 0;
+ while(fread(buf,sizeof(short),N,fin)) {
+ printf("%d\n", frames++);
+
+ /* Update input speech buffers */
+
+ for(i=0; i<M-N; i++)
+ Sn[i] = Sn[i+N];
+ for(i=0; i<N; i++)
+ Sn[i+M-N] = buf[i];
+ dft_speech(Sw, Sn, w);
+ dump_Sn(Sn); dump_Sw(Sw);
+
+ nlp(nlp_states,Sn,N,M,PITCH_MIN,PITCH_MAX,&pitch,Sw,&prev_Wo);
+ prev_Wo = TWO_PI/pitch;
+
+ fprintf(fout,"%f\n",pitch);
+ }
+
+ fclose(fin);
+ fclose(fout);
+ if (dump) dump_off();
+ nlp_destroy(nlp_states);
+
+ return 0;
+}
+
+
--- /dev/null
+/*---------------------------------------------------------------------------*\
+
+ FILE........: tquant.c
+ AUTHOR......: David Rowe
+ DATE CREATED: 22/8/10
+
+ Generates quantisation curves for plotting on Octave.
+
+\*---------------------------------------------------------------------------*/
+
+/*
+ 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, 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 <stdlib.h>
+#include <stdio.h>
+#include <string.h>
+#include <math.h>
+
+#include "defines.h"
+#include "dump.h"
+#include "quantise.h"
+
+int test_Wo_quant();
+int test_lsp_quant();
+int test_lsp(int lsp_number, int levels, float max_error_hz);
+int test_energy_quant(int levels, float max_error_dB);
+
+int main() {
+ quantise_init();
+ test_Wo_quant();
+ test_lsp_quant();
+ test_energy_quant(E_LEVELS, 0.5*(E_MAX_DB - E_MIN_DB)/E_LEVELS);
+
+ return 0;
+}
+
+int test_lsp_quant() {
+ test_lsp( 1, 16, 12.5);
+ test_lsp( 2, 16, 12.5);
+ test_lsp( 3, 16, 25);
+ test_lsp( 4, 16, 50);
+ test_lsp( 5, 16, 50);
+ test_lsp( 6, 16, 50);
+ test_lsp( 7, 16, 50);
+ test_lsp( 8, 8, 50);
+ test_lsp( 9, 8, 50);
+ test_lsp(10, 4, 100);
+
+ return 0;
+}
+
+int test_energy_quant(int levels, float max_error_dB) {
+ FILE *fe;
+ float e,e_dec, error, low_e, high_e;
+ int index, index_in, index_out, i;
+
+ /* check 1:1 match between input and output levels */
+
+ for(i=0; i<levels; i++) {
+ index_in = i;
+ e = decode_energy(index_in);
+ index_out = encode_energy(e);
+ if (index_in != index_out) {
+ printf("edB: %f index_in: %d index_out: %d\n",
+ 10.0*log10(e), index_in, index_out);
+ exit(0);
+ }
+ }
+
+ /* check error over range of quantiser */
+
+ low_e = decode_energy(0);
+ high_e = decode_energy(levels-1);
+ fe = fopen("energy_err.txt", "wt");
+
+ for(e=low_e; e<high_e; e +=(high_e-low_e)/1000.0) {
+ index = encode_energy(e);
+ e_dec = decode_energy(index);
+ error = 10.0*log10(e) - 10.0*log10(e_dec);
+ fprintf(fe, "%f\n", error);
+ if (fabs(error) > max_error_dB) {
+ printf("error: %f %f\n", error, max_error_dB);
+ exit(0);
+ }
+ }
+
+ fclose(fe);
+ return 0;
+}
+
+int test_lsp(int lsp_number, int levels, float max_error_hz) {
+ float lsp[LPC_ORD];
+ int indexes_in[LPC_ORD];
+ int indexes_out[LPC_ORD];
+ int indexes[LPC_ORD];
+ int i;
+ float lowf, highf, f, error;
+ char s[MAX_STR];
+ FILE *flsp;
+ float max_error_rads;
+
+ lsp_number--;
+ max_error_rads = max_error_hz*TWO_PI/FS;
+
+ for(i=0; i<LPC_ORD; i++)
+ indexes_in[i] = 0;
+
+ for(i=0; i<levels; i++) {
+ indexes_in[lsp_number] = i;
+ decode_lsps(lsp, indexes_in, LPC_ORD);
+ encode_lsps(indexes_out, lsp,LPC_ORD);
+ if (indexes_in[lsp_number] != indexes_out[lsp_number]) {
+ printf("freq: %f index_in: %d index_out: %d\n",
+ lsp[lsp_number]+1, indexes_in[lsp_number],
+ indexes_out[lsp_number]);
+ exit(0);
+ }
+ }
+
+ for(i=0; i<LPC_ORD; i++)
+ indexes[i] = 0;
+ indexes[lsp_number] = 0;
+ decode_lsps(lsp, indexes, LPC_ORD);
+ lowf = lsp[lsp_number];
+ indexes[lsp_number] = levels - 1;
+ decode_lsps(lsp, indexes, LPC_ORD);
+ highf = lsp[lsp_number];
+ sprintf(s,"lsp%d_err.txt", lsp_number+1);
+ flsp = fopen(s, "wt");
+
+ for(f=lowf; f<highf; f +=(highf-lowf)/1000.0) {
+ lsp[lsp_number] = f;
+ encode_lsps(indexes, lsp, LPC_ORD);
+ decode_lsps(lsp, indexes, LPC_ORD);
+ error = f - lsp[lsp_number];
+ fprintf(flsp, "%f\n", error);
+ if (fabs(error) > max_error_rads) {
+ printf("%d error: %f %f\n", lsp_number+1, error, max_error_rads);
+ exit(0);
+ }
+ }
+
+ fclose(flsp);
+
+ printf("OK\n");
+
+ return 0;
+}
+
+int test_Wo_quant() {
+ int c;
+ FILE *f;
+ float Wo,Wo_dec, error, step_size;
+ int index, index_in, index_out;
+
+ /* output Wo quant curve for plotting */
+
+ f = fopen("quant_pitch.txt","wt");
+
+ for(Wo=0.9*(TWO_PI/P_MAX); Wo<=1.1*(TWO_PI/P_MIN); Wo += 0.001) {
+ index = encode_Wo(Wo);
+ fprintf(f, "%f %d\n", Wo, index);
+ }
+
+ fclose(f);
+
+ /* check for all Wo codes we get 1:1 match between encoder
+ and decoder Wo levels */
+
+ for(c=0; c<WO_LEVELS; c++) {
+ index_in = c;
+ Wo = decode_Wo(index_in);
+ index_out = encode_Wo(Wo);
+ if (index_in != index_out)
+ printf(" Wo %f index_in %d index_out %d\n", Wo,
+ index_in, index_out);
+ }
+
+ /* measure quantisation error stats and compare to expected. Also
+ plot histogram of error file to check. */
+
+ f = fopen("quant_pitch_err.txt","wt");
+ step_size = ((TWO_PI/P_MIN) - (TWO_PI/P_MAX))/WO_LEVELS;
+
+ for(Wo=TWO_PI/P_MAX; Wo<0.99*TWO_PI/P_MIN; Wo += 0.0001) {
+ index = encode_Wo(Wo);
+ Wo_dec = decode_Wo(index);
+ error = Wo - Wo_dec;
+ if (fabs(error) > (step_size/2.0)) {
+ printf("error: %f step_size/2: %f\n", error, step_size/2.0);
+ exit(0);
+ }
+ fprintf(f,"%f\n",error);
+ }
+ printf("OK\n");
+
+ fclose(f);
+ return 0;
+}
--- /dev/null
+/*--------------------------------------------------------------------------*\
+
+ FILE........: VQTRAIN.C
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/2/95
+
+ This program trains vector quantisers using K dimensional Lloyd-Max
+ method.
+
+\*--------------------------------------------------------------------------*/
+
+/*
+ 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, 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.
+*/
+
+/*-----------------------------------------------------------------------*\
+
+ INCLUDES
+
+\*-----------------------------------------------------------------------*/
+
+#include <stdio.h>
+#include <stdlib.h>
+#include <string.h>
+#include <math.h>
+#include <ctype.h>
+
+/*-----------------------------------------------------------------------*\
+
+ DEFINES
+
+\*-----------------------------------------------------------------------*/
+
+#define DELTAQ 0.01 /* quiting distortion */
+#define MAX_STR 80 /* maximum string length */
+
+/*-----------------------------------------------------------------------*\
+
+ FUNCTION PROTOTYPES
+
+\*-----------------------------------------------------------------------*/
+
+void zero(float v[], int k);
+void acc(float v1[], float v2[], int k);
+void norm(float v[], int k, long n);
+long quantise(float cb[], float vec[], int k, int m, float *se);
+
+/*-----------------------------------------------------------------------*\
+
+ MAIN
+
+\*-----------------------------------------------------------------------*/
+
+int main(int argc, char *argv[]) {
+ long k,m; /* dimension and codebook size */
+ float *vec; /* current vector */
+ float *cb; /* vector codebook */
+ float *cent; /* centroids for each codebook entry */
+ long *n; /* number of vectors in this interval */
+ long J; /* number of vectors in training set */
+ long i,j;
+ long ind; /* index of current vector */
+ float se; /* squared error for this iteration */
+ float Dn,Dn_1; /* current and previous iterations distortion */
+ float delta; /* improvement in distortion */
+ FILE *ftrain; /* file containing training set */
+ FILE *fvq; /* file containing vector quantiser */
+
+ /* Interpret command line arguments */
+
+ if (argc != 5) {
+ printf("usage: vqtrain TrainFile K M VQFile\n");
+ exit(0);
+ }
+
+ /* Open training file */
+
+ ftrain = fopen(argv[1],"rb");
+ if (ftrain == NULL) {
+ printf("Error opening training database file: %s\n",argv[1]);
+ exit(1);
+ }
+
+ /* determine k and m, and allocate arrays */
+
+ k = atol(argv[2]);
+ m = atol(argv[3]);
+ printf("dimension K=%ld number of entries M=%ld\n", k,m);
+ vec = (float*)malloc(sizeof(float)*k);
+ cb = (float*)malloc(sizeof(float)*k*m);
+ cent = (float*)malloc(sizeof(float)*k*m);
+ n = (long*)malloc(sizeof(long)*m);
+ if (cb == NULL || cb == NULL || cent == NULL || vec == NULL) {
+ printf("Error in malloc.\n");
+ exit(1);
+ }
+
+ /* determine size of training set */
+
+ J = 0;
+ while(fread(vec, sizeof(float), k, ftrain) == k)
+ J++;
+ printf("J=%ld entries in training set\n", J);
+
+ /* set up initial codebook state from samples of training set */
+
+ rewind(ftrain);
+ fread(cb, sizeof(float), k*m, ftrain);
+
+ /* main loop */
+
+ Dn = 1E32;
+ j = 1;
+ do {
+ Dn_1 = Dn;
+
+ /* zero centroids */
+
+ for(i=0; i<m; i++) {
+ zero(¢[i*k], k);
+ n[i] = 0;
+ }
+
+ /* quantise training set */
+
+ se = 0.0;
+ rewind(ftrain);
+ for(i=0; i<J; i++) {
+ fread(vec, sizeof(float), k, ftrain);
+ ind = quantise(cb, vec, k, m, &se);
+ n[ind]++;
+ acc(¢[ind*k], vec, k);
+ }
+ Dn = se/J;
+ delta = (Dn_1-Dn)/Dn;
+
+ printf("\r Iteration %ld, Dn = %f, Delta = %e\n", j, Dn, delta);
+ j++;
+
+ /* determine new codebook from centriods */
+
+ if (delta > DELTAQ)
+ for(i=0; i<m; i++) {
+ if (n[i] != 0) {
+ norm(¢[i*k], k, n[i]);
+ memcpy(&cb[i*k], ¢[i*k], k*sizeof(float));
+ }
+ }
+
+ } while (delta > DELTAQ);
+
+ /* save codebook to disk */
+
+ fvq = fopen(argv[4],"wt");
+ if (fvq == NULL) {
+ printf("Error opening VQ file: %s\n",argv[4]);
+ exit(1);
+ }
+
+ for(j=0; j<m; j++) {
+ for(i=0; i<k; i++)
+ fprintf(fvq,"%f ",cb[j*k+i]);
+ fprintf(fvq,"\n");
+ }
+ fclose(fvq);
+
+ return 0;
+}
+
+/*-----------------------------------------------------------------------*\
+
+ FUNCTIONS
+
+\*-----------------------------------------------------------------------*/
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: zero()
+
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/2/95
+
+ Zeros a vector of length k.
+
+\*---------------------------------------------------------------------------*/
+
+void zero(float v[], int k)
+/* float v[]; ptr to start of vector */
+/* int k; lngth of vector */
+{
+ int i;
+
+ for(i=0; i<k; i++)
+ v[i] = 0.0;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: acc()
+
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/2/95
+
+ Adds k dimensional vectors v1 to v2 and stores the result back in v1.
+
+\*---------------------------------------------------------------------------*/
+
+void acc(float v1[], float v2[], int k)
+/* float v1[]; ptr to start of vector to accumulate */
+/* float v2[]; ptr to start of vector to add */
+/* int k; dimension of vectors */
+{
+ int i;
+
+ for(i=0; i<k; i++)
+ v1[i] += v2[i];
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: norm()
+
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/2/95
+
+ Divides each element in k dimensional vector v by n.
+
+\*---------------------------------------------------------------------------*/
+
+void norm(float v[], int k, long n)
+/* float v[]; ptr to start of vector */
+/* int k; dimension of vectors */
+/* int n; normalising factor */
+{
+ int i;
+
+ for(i=0; i<k; i++)
+ v[i] /= n;
+}
+
+/*---------------------------------------------------------------------------*\
+
+ FUNCTION....: quantise()
+
+ AUTHOR......: David Rowe
+ DATE CREATED: 23/2/95
+
+ Quantises vec by choosing the nearest vector in codebook cb, and
+ returns the vector index. The squared error of the quantised vector
+ is added to se.
+
+\*---------------------------------------------------------------------------*/
+
+long quantise(float cb[], float vec[], int k, int m, float *se)
+/* float cb[][K]; current VQ codebook */
+/* float vec[]; vector to quantise */
+/* int k; dimension of vectors */
+/* int m; size of codebook */
+/* float *se; accumulated squared error */
+{
+ float e; /* current error */
+ long besti; /* best index so far */
+ float beste; /* best error so far */
+ long j;
+ int i;
+
+ besti = 0;
+ beste = 1E32;
+ for(j=0; j<m; j++) {
+ e = 0.0;
+ for(i=0; i<k; i++)
+ e += pow(cb[j*k+i]-vec[i],2.0);
+ if (e < beste) {
+ beste = e;
+ besti = j;
+ }
+ }
+
+ *se += beste;
+
+ return(besti);
+}
+
BASE=../../../..
-CODEC2_DIR=$(switch_srcdir)/libs/libcodec2-1.0
-CODEC2_BUILDDIR=$(switch_builddir)/libs/libcodec2-1.0
+CODEC2_DIR=$(switch_srcdir)/libs/libcodec2
+CODEC2_BUILDDIR=$(switch_builddir)/libs/libcodec2
CODEC2LA=$(CODEC2_BUILDDIR)/src/libcodec2.la
LOCAL_CFLAGS=-I$(CODEC2_DIR)/include -I$(CODEC2_BUILDDIR)/src
projects / thirdparty / freeswitch.git / commitdiff
