13 files changed, 4133 insertions, 0 deletions
diff --git a/ext/jack/Makefile.am b/ext/jack/Makefile.am
new file mode 100644
index 0000000..e786a8d
--- /dev/null
+++ b/ext/jack/Makefile.am
@@ -0,0 +1,10 @@
+
+plugin_LTLIBRARIES = libgstjack.la
+
+libgstjack_la_SOURCES = gstjackutil.c gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c
+libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS)
+libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(JACK_LIBS)
+libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstjack_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
+
+noinst_HEADERS = gstjackutil.h gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h
diff --git a/ext/jack/Makefile.in b/ext/jack/Makefile.in
new file mode 100755
index 0000000..0ecf697
--- /dev/null
+++ b/ext/jack/Makefile.in
@@ -0,0 +1,926 @@
+# Makefile.in generated by automake 1.14.1 from Makefile.am.
+# @configure_input@
+
+# Copyright (C) 1994-2013 Free Software Foundation, Inc.
+
+# This Makefile.in is free software; the Free Software Foundation
+# gives unlimited permission to copy and/or distribute it,
+# with or without modifications, as long as this notice is preserved.
+
+# This program is distributed in the hope that it will be useful,
+# but WITHOUT ANY WARRANTY, to the extent permitted by law; without
+# even the implied warranty of MERCHANTABILITY or FITNESS FOR A
+# PARTICULAR PURPOSE.
+
+@SET_MAKE@
+
+
+VPATH = @srcdir@
+am__is_gnu_make = test -n '$(MAKEFILE_LIST)' && test -n '$(MAKELEVEL)'
+am__make_running_with_option = \
+  case $${target_option-} in \
+      ?) ;; \
+      *) echo "am__make_running_with_option: internal error: invalid" \
+              "target option '$${target_option-}' specified" >&2; \
+         exit 1;; \
+  esac; \
+  has_opt=no; \
+  sane_makeflags=$$MAKEFLAGS; \
+  if $(am__is_gnu_make); then \
+    sane_makeflags=$$MFLAGS; \
+  else \
+    case $$MAKEFLAGS in \
+      *\\[\ \	]*) \
+        bs=\\; \
+        sane_makeflags=`printf '%s\n' "$$MAKEFLAGS" \
+          | sed "s/$$bs$$bs[$$bs $$bs	]*//g"`;; \
+    esac; \
+  fi; \
+  skip_next=no; \
+  strip_trailopt () \
+  { \
+    flg=`printf '%s\n' "$$flg" | sed "s/$$1.*$$//"`; \
+  }; \
+  for flg in $$sane_makeflags; do \
+    test $$skip_next = yes && { skip_next=no; continue; }; \
+    case $$flg in \
+      *=*|--*) continue;; \
+        -*I) strip_trailopt 'I'; skip_next=yes;; \
+      -*I?*) strip_trailopt 'I';; \
+        -*O) strip_trailopt 'O'; skip_next=yes;; \
+      -*O?*) strip_trailopt 'O';; \
+        -*l) strip_trailopt 'l'; skip_next=yes;; \
+      -*l?*) strip_trailopt 'l';; \
+      -[dEDm]) skip_next=yes;; \
+      -[JT]) skip_next=yes;; \
+    esac; \
+    case $$flg in \
+      *$$target_option*) has_opt=yes; break;; \
+    esac; \
+  done; \
+  test $$has_opt = yes
+am__make_dryrun = (target_option=n; $(am__make_running_with_option))
+am__make_keepgoing = (target_option=k; $(am__make_running_with_option))
+pkgdatadir = $(datadir)/@PACKAGE@
+pkgincludedir = $(includedir)/@PACKAGE@
+pkglibdir = $(libdir)/@PACKAGE@
+pkglibexecdir = $(libexecdir)/@PACKAGE@
+am__cd = CDPATH="$${ZSH_VERSION+.}$(PATH_SEPARATOR)" && cd
+install_sh_DATA = $(install_sh) -c -m 644
+install_sh_PROGRAM = $(install_sh) -c
+install_sh_SCRIPT = $(install_sh) -c
+INSTALL_HEADER = $(INSTALL_DATA)
+transform = $(program_transform_name)
+NORMAL_INSTALL = :
+PRE_INSTALL = :
+POST_INSTALL = :
+NORMAL_UNINSTALL = :
+PRE_UNINSTALL = :
+POST_UNINSTALL = :
+build_triplet = @build@
+host_triplet = @host@
+target_triplet = @target@
+subdir = ext/jack
+DIST_COMMON = $(srcdir)/Makefile.in $(srcdir)/Makefile.am \
+	$(top_srcdir)/depcomp $(noinst_HEADERS)
+ACLOCAL_M4 = $(top_srcdir)/aclocal.m4
+am__aclocal_m4_deps = $(top_srcdir)/common/m4/as-ac-expand.m4 \
+	$(top_srcdir)/common/m4/as-auto-alt.m4 \
+	$(top_srcdir)/common/m4/as-compiler-flag.m4 \
+	$(top_srcdir)/common/m4/as-gcc-inline-assembly.m4 \
+	$(top_srcdir)/common/m4/as-libtool.m4 \
+	$(top_srcdir)/common/m4/as-version.m4 \
+	$(top_srcdir)/common/m4/ax_create_stdint_h.m4 \
+	$(top_srcdir)/common/m4/gst-arch.m4 \
+	$(top_srcdir)/common/m4/gst-args.m4 \
+	$(top_srcdir)/common/m4/gst-check.m4 \
+	$(top_srcdir)/common/m4/gst-default.m4 \
+	$(top_srcdir)/common/m4/gst-dowhile.m4 \
+	$(top_srcdir)/common/m4/gst-error.m4 \
+	$(top_srcdir)/common/m4/gst-feature.m4 \
+	$(top_srcdir)/common/m4/gst-gettext.m4 \
+	$(top_srcdir)/common/m4/gst-glib2.m4 \
+	$(top_srcdir)/common/m4/gst-package-release-datetime.m4 \
+	$(top_srcdir)/common/m4/gst-platform.m4 \
+	$(top_srcdir)/common/m4/gst-plugin-docs.m4 \
+	$(top_srcdir)/common/m4/gst-plugindir.m4 \
+	$(top_srcdir)/common/m4/gst-x11.m4 \
+	$(top_srcdir)/common/m4/gst.m4 \
+	$(top_srcdir)/common/m4/gtk-doc.m4 \
+	$(top_srcdir)/common/m4/orc.m4 $(top_srcdir)/common/m4/pkg.m4 \
+	$(top_srcdir)/m4/aalib.m4 $(top_srcdir)/m4/gettext.m4 \
+	$(top_srcdir)/m4/gst-fionread.m4 $(top_srcdir)/m4/iconv.m4 \
+	$(top_srcdir)/m4/intlmacosx.m4 $(top_srcdir)/m4/lib-ld.m4 \
+	$(top_srcdir)/m4/lib-link.m4 $(top_srcdir)/m4/lib-prefix.m4 \
+	$(top_srcdir)/m4/libtool.m4 $(top_srcdir)/m4/ltoptions.m4 \
+	$(top_srcdir)/m4/ltsugar.m4 $(top_srcdir)/m4/ltversion.m4 \
+	$(top_srcdir)/m4/lt~obsolete.m4 $(top_srcdir)/m4/nls.m4 \
+	$(top_srcdir)/m4/po.m4 $(top_srcdir)/m4/progtest.m4 \
+	$(top_srcdir)/configure.ac
+am__configure_deps = $(am__aclocal_m4_deps) $(CONFIGURE_DEPENDENCIES) \
+	$(ACLOCAL_M4)
+mkinstalldirs = $(install_sh) -d
+CONFIG_HEADER = $(top_builddir)/config.h
+CONFIG_CLEAN_FILES =
+CONFIG_CLEAN_VPATH_FILES =
+am__vpath_adj_setup = srcdirstrip=`echo "$(srcdir)" | sed 's|.|.|g'`;
+am__vpath_adj = case $$p in \
+    $(srcdir)/*) f=`echo "$$p" | sed "s|^$$srcdirstrip/||"`;; \
+    *) f=$$p;; \
+  esac;
+am__strip_dir = f=`echo $$p | sed -e 's|^.*/||'`;
+am__install_max = 40
+am__nobase_strip_setup = \
+  srcdirstrip=`echo "$(srcdir)" | sed 's/[].[^$$\\*|]/\\\\&/g'`
+am__nobase_strip = \
+  for p in $$list; do echo "$$p"; done | sed -e "s|$$srcdirstrip/||"
+am__nobase_list = $(am__nobase_strip_setup); \
+  for p in $$list; do echo "$$p $$p"; done | \
+  sed "s| $$srcdirstrip/| |;"' / .*\//!s/ .*/ ./; s,\( .*\)/[^/]*$$,\1,' | \
+  $(AWK) 'BEGIN { files["."] = "" } { files[$$2] = files[$$2] " " $$1; \
+    if (++n[$$2] == $(am__install_max)) \
+      { print $$2, files[$$2]; n[$$2] = 0; files[$$2] = "" } } \
+    END { for (dir in files) print dir, files[dir] }'
+am__base_list = \
+  sed '$$!N;$$!N;$$!N;$$!N;$$!N;$$!N;$$!N;s/\n/ /g' | \
+  sed '$$!N;$$!N;$$!N;$$!N;s/\n/ /g'
+am__uninstall_files_from_dir = { \
+  test -z "$$files" \
+    || { test ! -d "$$dir" && test ! -f "$$dir" && test ! -r "$$dir"; } \
+    || { echo " ( cd '$$dir' && rm -f" $$files ")"; \
+         $(am__cd) "$$dir" && rm -f $$files; }; \
+  }
+am__installdirs = "$(DESTDIR)$(plugindir)"
+LTLIBRARIES = $(plugin_LTLIBRARIES)
+am__DEPENDENCIES_1 =
+libgstjack_la_DEPENDENCIES = $(am__DEPENDENCIES_1) \
+	$(am__DEPENDENCIES_1)
+am_libgstjack_la_OBJECTS = libgstjack_la-gstjackutil.lo \
+	libgstjack_la-gstjack.lo libgstjack_la-gstjackaudiosrc.lo \
+	libgstjack_la-gstjackaudiosink.lo \
+	libgstjack_la-gstjackaudioclient.lo
+libgstjack_la_OBJECTS = $(am_libgstjack_la_OBJECTS)
+AM_V_lt = $(am__v_lt_@AM_V@)
+am__v_lt_ = $(am__v_lt_@AM_DEFAULT_V@)
+am__v_lt_0 = --silent
+am__v_lt_1 = 
+libgstjack_la_LINK = $(LIBTOOL) $(AM_V_lt) --tag=CC \
+	$(libgstjack_la_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=link \
+	$(CCLD) $(libgstjack_la_CFLAGS) $(CFLAGS) \
+	$(libgstjack_la_LDFLAGS) $(LDFLAGS) -o $@
+AM_V_P = $(am__v_P_@AM_V@)
+am__v_P_ = $(am__v_P_@AM_DEFAULT_V@)
+am__v_P_0 = false
+am__v_P_1 = :
+AM_V_GEN = $(am__v_GEN_@AM_V@)
+am__v_GEN_ = $(am__v_GEN_@AM_DEFAULT_V@)
+am__v_GEN_0 = @echo "  GEN     " $@;
+am__v_GEN_1 = 
+AM_V_at = $(am__v_at_@AM_V@)
+am__v_at_ = $(am__v_at_@AM_DEFAULT_V@)
+am__v_at_0 = @
+am__v_at_1 = 
+DEFAULT_INCLUDES = -I.@am__isrc@ -I$(top_builddir)
+depcomp = $(SHELL) $(top_srcdir)/depcomp
+am__depfiles_maybe = depfiles
+am__mv = mv -f
+COMPILE = $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) \
+	$(CPPFLAGS) $(AM_CFLAGS) $(CFLAGS)
+LTCOMPILE = $(LIBTOOL) $(AM_V_lt) --tag=CC $(AM_LIBTOOLFLAGS) \
+	$(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) \
+	$(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) \
+	$(AM_CFLAGS) $(CFLAGS)
+AM_V_CC = $(am__v_CC_@AM_V@)
+am__v_CC_ = $(am__v_CC_@AM_DEFAULT_V@)
+am__v_CC_0 = @echo "  CC      " $@;
+am__v_CC_1 = 
+CCLD = $(CC)
+LINK = $(LIBTOOL) $(AM_V_lt) --tag=CC $(AM_LIBTOOLFLAGS) \
+	$(LIBTOOLFLAGS) --mode=link $(CCLD) $(AM_CFLAGS) $(CFLAGS) \
+	$(AM_LDFLAGS) $(LDFLAGS) -o $@
+AM_V_CCLD = $(am__v_CCLD_@AM_V@)
+am__v_CCLD_ = $(am__v_CCLD_@AM_DEFAULT_V@)
+am__v_CCLD_0 = @echo "  CCLD    " $@;
+am__v_CCLD_1 = 
+SOURCES = $(libgstjack_la_SOURCES)
+DIST_SOURCES = $(libgstjack_la_SOURCES)
+am__can_run_installinfo = \
+  case $$AM_UPDATE_INFO_DIR in \
+    n|no|NO) false;; \
+    *) (install-info --version) >/dev/null 2>&1;; \
+  esac
+HEADERS = $(noinst_HEADERS)
+am__tagged_files = $(HEADERS) $(SOURCES) $(TAGS_FILES) $(LISP)
+# Read a list of newline-separated strings from the standard input,
+# and print each of them once, without duplicates.  Input order is
+# *not* preserved.
+am__uniquify_input = $(AWK) '\
+  BEGIN { nonempty = 0; } \
+  { items[$$0] = 1; nonempty = 1; } \
+  END { if (nonempty) { for (i in items) print i; }; } \
+'
+# Make sure the list of sources is unique.  This is necessary because,
+# e.g., the same source file might be shared among _SOURCES variables
+# for different programs/libraries.
+am__define_uniq_tagged_files = \
+  list='$(am__tagged_files)'; \
+  unique=`for i in $$list; do \
+    if test -f "$$i"; then echo $$i; else echo $(srcdir)/$$i; fi; \
+  done | $(am__uniquify_input)`
+ETAGS = etags
+CTAGS = ctags
+DISTFILES = $(DIST_COMMON) $(DIST_SOURCES) $(TEXINFOS) $(EXTRA_DIST)
+AALIB_CFLAGS = @AALIB_CFLAGS@
+AALIB_CONFIG = @AALIB_CONFIG@
+AALIB_LIBS = @AALIB_LIBS@
+ACLOCAL = @ACLOCAL@
+ACLOCAL_AMFLAGS = @ACLOCAL_AMFLAGS@
+AMTAR = @AMTAR@
+AM_DEFAULT_VERBOSITY = @AM_DEFAULT_VERBOSITY@
+AR = @AR@
+AS = @AS@
+AUTOCONF = @AUTOCONF@
+AUTOHEADER = @AUTOHEADER@
+AUTOMAKE = @AUTOMAKE@
+AWK = @AWK@
+BZ2_LIBS = @BZ2_LIBS@
+CAIRO_CFLAGS = @CAIRO_CFLAGS@
+CAIRO_LIBS = @CAIRO_LIBS@
+CC = @CC@
+CCAS = @CCAS@
+CCASDEPMODE = @CCASDEPMODE@
+CCASFLAGS = @CCASFLAGS@
+CCDEPMODE = @CCDEPMODE@
+CFLAGS = @CFLAGS@
+CPP = @CPP@
+CPPFLAGS = @CPPFLAGS@
+CXX = @CXX@
+CXXCPP = @CXXCPP@
+CXXDEPMODE = @CXXDEPMODE@
+CXXFLAGS = @CXXFLAGS@
+CYGPATH_W = @CYGPATH_W@
+DEFAULT_AUDIOSINK = @DEFAULT_AUDIOSINK@
+DEFAULT_AUDIOSRC = @DEFAULT_AUDIOSRC@
+DEFAULT_VIDEOSINK = @DEFAULT_VIDEOSINK@
+DEFAULT_VIDEOSRC = @DEFAULT_VIDEOSRC@
+DEFAULT_VISUALIZER = @DEFAULT_VISUALIZER@
+DEFS = @DEFS@
+DEPDIR = @DEPDIR@
+DEPRECATED_CFLAGS = @DEPRECATED_CFLAGS@
+DIRECTSOUND_CFLAGS = @DIRECTSOUND_CFLAGS@
+DIRECTSOUND_LDFLAGS = @DIRECTSOUND_LDFLAGS@
+DIRECTSOUND_LIBS = @DIRECTSOUND_LIBS@
+DLLTOOL = @DLLTOOL@
+DSYMUTIL = @DSYMUTIL@
+DUMPBIN = @DUMPBIN@
+DV1394_CFLAGS = @DV1394_CFLAGS@
+DV1394_LIBS = @DV1394_LIBS@
+ECHO_C = @ECHO_C@
+ECHO_N = @ECHO_N@
+ECHO_T = @ECHO_T@
+EGREP = @EGREP@
+ERROR_CFLAGS = @ERROR_CFLAGS@
+ERROR_CXXFLAGS = @ERROR_CXXFLAGS@
+ERROR_OBJCFLAGS = @ERROR_OBJCFLAGS@
+EXEEXT = @EXEEXT@
+FFLAGS = @FFLAGS@
+FGREP = @FGREP@
+FLAC_CFLAGS = @FLAC_CFLAGS@
+FLAC_LIBS = @FLAC_LIBS@
+GCOV = @GCOV@
+GCOV_CFLAGS = @GCOV_CFLAGS@
+GCOV_LIBS = @GCOV_LIBS@
+GDK_PIXBUF_CFLAGS = @GDK_PIXBUF_CFLAGS@
+GDK_PIXBUF_LIBS = @GDK_PIXBUF_LIBS@
+GETTEXT_MACRO_VERSION = @GETTEXT_MACRO_VERSION@
+GETTEXT_PACKAGE = @GETTEXT_PACKAGE@
+GIO_CFLAGS = @GIO_CFLAGS@
+GIO_LDFLAGS = @GIO_LDFLAGS@
+GIO_LIBS = @GIO_LIBS@
+GLIB_CFLAGS = @GLIB_CFLAGS@
+GLIB_EXTRA_CFLAGS = @GLIB_EXTRA_CFLAGS@
+GLIB_GENMARSHAL = @GLIB_GENMARSHAL@
+GLIB_LIBS = @GLIB_LIBS@
+GLIB_MKENUMS = @GLIB_MKENUMS@
+GLIB_PREFIX = @GLIB_PREFIX@
+GLIB_REQ = @GLIB_REQ@
+GMSGFMT = @GMSGFMT@
+GMSGFMT_015 = @GMSGFMT_015@
+GREP = @GREP@
+GSTPB_PLUGINS_DIR = @GSTPB_PLUGINS_DIR@
+GSTPB_PREFIX = @GSTPB_PREFIX@
+GST_AGE = @GST_AGE@
+GST_ALL_LDFLAGS = @GST_ALL_LDFLAGS@
+GST_API_VERSION = @GST_API_VERSION@
+GST_BASE_CFLAGS = @GST_BASE_CFLAGS@
+GST_BASE_LIBS = @GST_BASE_LIBS@
+GST_CFLAGS = @GST_CFLAGS@
+GST_CHECK_CFLAGS = @GST_CHECK_CFLAGS@
+GST_CHECK_LIBS = @GST_CHECK_LIBS@
+GST_CONTROLLER_CFLAGS = @GST_CONTROLLER_CFLAGS@
+GST_CONTROLLER_LIBS = @GST_CONTROLLER_LIBS@
+GST_CURRENT = @GST_CURRENT@
+GST_CXXFLAGS = @GST_CXXFLAGS@
+GST_LEVEL_DEFAULT = @GST_LEVEL_DEFAULT@
+GST_LIBS = @GST_LIBS@
+GST_LIBVERSION = @GST_LIBVERSION@
+GST_LICENSE = @GST_LICENSE@
+GST_LT_LDFLAGS = @GST_LT_LDFLAGS@
+GST_NET_CFLAGS = @GST_NET_CFLAGS@
+GST_NET_LIBS = @GST_NET_LIBS@
+GST_OBJCFLAGS = @GST_OBJCFLAGS@
+GST_OPTION_CFLAGS = @GST_OPTION_CFLAGS@
+GST_OPTION_CXXFLAGS = @GST_OPTION_CXXFLAGS@
+GST_OPTION_OBJCFLAGS = @GST_OPTION_OBJCFLAGS@
+GST_PACKAGE_NAME = @GST_PACKAGE_NAME@
+GST_PACKAGE_ORIGIN = @GST_PACKAGE_ORIGIN@
+GST_PLUGINS_ALL = @GST_PLUGINS_ALL@
+GST_PLUGINS_BASE_CFLAGS = @GST_PLUGINS_BASE_CFLAGS@
+GST_PLUGINS_BASE_DIR = @GST_PLUGINS_BASE_DIR@
+GST_PLUGINS_BASE_LIBS = @GST_PLUGINS_BASE_LIBS@
+GST_PLUGINS_DIR = @GST_PLUGINS_DIR@
+GST_PLUGINS_NONPORTED = @GST_PLUGINS_NONPORTED@
+GST_PLUGINS_SELECTED = @GST_PLUGINS_SELECTED@
+GST_PLUGIN_LDFLAGS = @GST_PLUGIN_LDFLAGS@
+GST_PLUGIN_LIBTOOLFLAGS = @GST_PLUGIN_LIBTOOLFLAGS@
+GST_PREFIX = @GST_PREFIX@
+GST_REVISION = @GST_REVISION@
+GST_TOOLS_DIR = @GST_TOOLS_DIR@
+GTKDOC_CHECK = @GTKDOC_CHECK@
+GTKDOC_DEPS_CFLAGS = @GTKDOC_DEPS_CFLAGS@
+GTKDOC_DEPS_LIBS = @GTKDOC_DEPS_LIBS@
+GTKDOC_MKPDF = @GTKDOC_MKPDF@
+GTKDOC_REBASE = @GTKDOC_REBASE@
+GTK_CFLAGS = @GTK_CFLAGS@
+GTK_LIBS = @GTK_LIBS@
+GTK_X11_CFLAGS = @GTK_X11_CFLAGS@
+GTK_X11_LIBS = @GTK_X11_LIBS@
+GUDEV_CFLAGS = @GUDEV_CFLAGS@
+GUDEV_LIBS = @GUDEV_LIBS@
+HAVE_AVC1394 = @HAVE_AVC1394@
+HAVE_CXX = @HAVE_CXX@
+HAVE_DIRECTSOUND = @HAVE_DIRECTSOUND@
+HAVE_ROM1394 = @HAVE_ROM1394@
+HAVE_SPEEX = @HAVE_SPEEX@
+HAVE_X = @HAVE_X@
+HAVE_XSHM = @HAVE_XSHM@
+HAVE_ZLIB = @HAVE_ZLIB@
+HTML_DIR = @HTML_DIR@
+INSTALL = @INSTALL@
+INSTALL_DATA = @INSTALL_DATA@
+INSTALL_PROGRAM = @INSTALL_PROGRAM@
+INSTALL_SCRIPT = @INSTALL_SCRIPT@
+INSTALL_STRIP_PROGRAM = @INSTALL_STRIP_PROGRAM@
+INTLLIBS = @INTLLIBS@
+INTL_MACOSX_LIBS = @INTL_MACOSX_LIBS@
+JACK_0_120_1_CFLAGS = @JACK_0_120_1_CFLAGS@
+JACK_0_120_1_LIBS = @JACK_0_120_1_LIBS@
+JACK_1_9_7_CFLAGS = @JACK_1_9_7_CFLAGS@
+JACK_1_9_7_LIBS = @JACK_1_9_7_LIBS@
+JACK_CFLAGS = @JACK_CFLAGS@
+JACK_LIBS = @JACK_LIBS@
+JPEG_LIBS = @JPEG_LIBS@
+LD = @LD@
+LDFLAGS = @LDFLAGS@
+LIBCACA_CFLAGS = @LIBCACA_CFLAGS@
+LIBCACA_LIBS = @LIBCACA_LIBS@
+LIBDV_CFLAGS = @LIBDV_CFLAGS@
+LIBDV_LIBS = @LIBDV_LIBS@
+LIBICONV = @LIBICONV@
+LIBIEC61883_CFLAGS = @LIBIEC61883_CFLAGS@
+LIBIEC61883_LIBS = @LIBIEC61883_LIBS@
+LIBINTL = @LIBINTL@
+LIBM = @LIBM@
+LIBOBJS = @LIBOBJS@
+LIBPNG_CFLAGS = @LIBPNG_CFLAGS@
+LIBPNG_LIBS = @LIBPNG_LIBS@
+LIBRT = @LIBRT@
+LIBS = @LIBS@
+LIBTOOL = @LIBTOOL@
+LIBV4L2_CFLAGS = @LIBV4L2_CFLAGS@
+LIBV4L2_LIBS = @LIBV4L2_LIBS@
+LIPO = @LIPO@
+LN_S = @LN_S@
+LOCALEDIR = @LOCALEDIR@
+LTLIBICONV = @LTLIBICONV@
+LTLIBINTL = @LTLIBINTL@
+LTLIBOBJS = @LTLIBOBJS@
+MAINT = @MAINT@
+MAKEINFO = @MAKEINFO@
+MANIFEST_TOOL = @MANIFEST_TOOL@
+MKDIR_P = @MKDIR_P@
+MSGFMT = @MSGFMT@
+MSGFMT_015 = @MSGFMT_015@
+MSGMERGE = @MSGMERGE@
+NM = @NM@
+NMEDIT = @NMEDIT@
+OBJC = @OBJC@
+OBJCDEPMODE = @OBJCDEPMODE@
+OBJCFLAGS = @OBJCFLAGS@
+OBJDUMP = @OBJDUMP@
+OBJEXT = @OBJEXT@
+ORCC = @ORCC@
+ORCC_FLAGS = @ORCC_FLAGS@
+ORC_CFLAGS = @ORC_CFLAGS@
+ORC_LIBS = @ORC_LIBS@
+OTOOL = @OTOOL@
+OTOOL64 = @OTOOL64@
+PACKAGE = @PACKAGE@
+PACKAGE_BUGREPORT = @PACKAGE_BUGREPORT@
+PACKAGE_NAME = @PACKAGE_NAME@
+PACKAGE_STRING = @PACKAGE_STRING@
+PACKAGE_TARNAME = @PACKAGE_TARNAME@
+PACKAGE_URL = @PACKAGE_URL@
+PACKAGE_VERSION = @PACKAGE_VERSION@
+PACKAGE_VERSION_MAJOR = @PACKAGE_VERSION_MAJOR@
+PACKAGE_VERSION_MICRO = @PACKAGE_VERSION_MICRO@
+PACKAGE_VERSION_MINOR = @PACKAGE_VERSION_MINOR@
+PACKAGE_VERSION_NANO = @PACKAGE_VERSION_NANO@
+PACKAGE_VERSION_RELEASE = @PACKAGE_VERSION_RELEASE@
+PATH_SEPARATOR = @PATH_SEPARATOR@
+PKG_CONFIG = @PKG_CONFIG@
+PLUGINDIR = @PLUGINDIR@
+POSUB = @POSUB@
+PROFILE_CFLAGS = @PROFILE_CFLAGS@
+PULSE_CFLAGS = @PULSE_CFLAGS@
+PULSE_LIBS = @PULSE_LIBS@
+PYTHON = @PYTHON@
+PYTHON_EXEC_PREFIX = @PYTHON_EXEC_PREFIX@
+PYTHON_PLATFORM = @PYTHON_PLATFORM@
+PYTHON_PREFIX = @PYTHON_PREFIX@
+PYTHON_VERSION = @PYTHON_VERSION@
+RANLIB = @RANLIB@
+RAW1394_CFLAGS = @RAW1394_CFLAGS@
+RAW1394_LIBS = @RAW1394_LIBS@
+SED = @SED@
+SET_MAKE = @SET_MAKE@
+SHELL = @SHELL@
+SHOUT2_CFLAGS = @SHOUT2_CFLAGS@
+SHOUT2_LIBS = @SHOUT2_LIBS@
+SOUP_CFLAGS = @SOUP_CFLAGS@
+SOUP_LIBS = @SOUP_LIBS@
+SPEEX_CFLAGS = @SPEEX_CFLAGS@
+SPEEX_LIBS = @SPEEX_LIBS@
+STRIP = @STRIP@
+TAGLIB_CFLAGS = @TAGLIB_CFLAGS@
+TAGLIB_CXXFLAGS = @TAGLIB_CXXFLAGS@
+TAGLIB_LIBS = @TAGLIB_LIBS@
+USE_NLS = @USE_NLS@
+VALGRIND_CFLAGS = @VALGRIND_CFLAGS@
+VALGRIND_LIBS = @VALGRIND_LIBS@
+VALGRIND_PATH = @VALGRIND_PATH@
+VERSION = @VERSION@
+VPX_130_CFLAGS = @VPX_130_CFLAGS@
+VPX_130_LIBS = @VPX_130_LIBS@
+VPX_CFLAGS = @VPX_CFLAGS@
+VPX_LIBS = @VPX_LIBS@
+WARNING_CFLAGS = @WARNING_CFLAGS@
+WARNING_CXXFLAGS = @WARNING_CXXFLAGS@
+WARNING_OBJCFLAGS = @WARNING_OBJCFLAGS@
+WAVPACK_CFLAGS = @WAVPACK_CFLAGS@
+WAVPACK_LIBS = @WAVPACK_LIBS@
+XDAMAGE_CFLAGS = @XDAMAGE_CFLAGS@
+XDAMAGE_LIBS = @XDAMAGE_LIBS@
+XFIXES_CFLAGS = @XFIXES_CFLAGS@
+XFIXES_LIBS = @XFIXES_LIBS@
+XGETTEXT = @XGETTEXT@
+XGETTEXT_015 = @XGETTEXT_015@
+XGETTEXT_EXTRA_OPTIONS = @XGETTEXT_EXTRA_OPTIONS@
+XMKMF = @XMKMF@
+XSHM_LIBS = @XSHM_LIBS@
+X_CFLAGS = @X_CFLAGS@
+X_EXTRA_LIBS = @X_EXTRA_LIBS@
+X_LIBS = @X_LIBS@
+X_PRE_LIBS = @X_PRE_LIBS@
+ZLIB_LIBS = @ZLIB_LIBS@
+abs_builddir = @abs_builddir@
+abs_srcdir = @abs_srcdir@
+abs_top_builddir = @abs_top_builddir@
+abs_top_srcdir = @abs_top_srcdir@
+ac_ct_AR = @ac_ct_AR@
+ac_ct_CC = @ac_ct_CC@
+ac_ct_CXX = @ac_ct_CXX@
+ac_ct_DUMPBIN = @ac_ct_DUMPBIN@
+ac_ct_OBJC = @ac_ct_OBJC@
+am__include = @am__include@
+am__leading_dot = @am__leading_dot@
+am__quote = @am__quote@
+am__tar = @am__tar@
+am__untar = @am__untar@
+bindir = @bindir@
+build = @build@
+build_alias = @build_alias@
+build_cpu = @build_cpu@
+build_os = @build_os@
+build_vendor = @build_vendor@
+builddir = @builddir@
+datadir = @datadir@
+datarootdir = @datarootdir@
+docdir = @docdir@
+dvidir = @dvidir@
+exec_prefix = @exec_prefix@
+host = @host@
+host_alias = @host_alias@
+host_cpu = @host_cpu@
+host_os = @host_os@
+host_vendor = @host_vendor@
+htmldir = @htmldir@
+includedir = @includedir@
+infodir = @infodir@
+install_sh = @install_sh@
+libdir = @libdir@
+libexecdir = @libexecdir@
+localedir = @localedir@
+localstatedir = @localstatedir@
+mandir = @mandir@
+mkdir_p = @mkdir_p@
+oldincludedir = @oldincludedir@
+pdfdir = @pdfdir@
+pkgpyexecdir = @pkgpyexecdir@
+pkgpythondir = @pkgpythondir@
+plugindir = @plugindir@
+prefix = @prefix@
+program_transform_name = @program_transform_name@
+psdir = @psdir@
+pyexecdir = @pyexecdir@
+pythondir = @pythondir@
+sbindir = @sbindir@
+sharedstatedir = @sharedstatedir@
+srcdir = @srcdir@
+sysconfdir = @sysconfdir@
+target = @target@
+target_alias = @target_alias@
+target_cpu = @target_cpu@
+target_os = @target_os@
+target_vendor = @target_vendor@
+top_build_prefix = @top_build_prefix@
+top_builddir = @top_builddir@
+top_srcdir = @top_srcdir@
+plugin_LTLIBRARIES = libgstjack.la
+libgstjack_la_SOURCES = gstjackutil.c gstjack.c gstjackaudiosrc.c gstjackaudiosink.c gstjackaudioclient.c
+libgstjack_la_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(GST_CFLAGS) $(JACK_CFLAGS)
+libgstjack_la_LIBADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_API_VERSION) $(JACK_LIBS)
+libgstjack_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstjack_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
+noinst_HEADERS = gstjackutil.h gstjackaudiosrc.h gstjackaudiosink.h gstjackaudioclient.h gstjack.h gstjackringbuffer.h
+all: all-am
+
+.SUFFIXES:
+.SUFFIXES: .c .lo .o .obj
+$(srcdir)/Makefile.in: @MAINTAINER_MODE_TRUE@ $(srcdir)/Makefile.am  $(am__configure_deps)
+	@for dep in $?; do \
+	  case '$(am__configure_deps)' in \
+	    *$$dep*) \
+	      ( cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh ) \
+	        && { if test -f $@; then exit 0; else break; fi; }; \
+	      exit 1;; \
+	  esac; \
+	done; \
+	echo ' cd $(top_srcdir) && $(AUTOMAKE) --gnu ext/jack/Makefile'; \
+	$(am__cd) $(top_srcdir) && \
+	  $(AUTOMAKE) --gnu ext/jack/Makefile
+.PRECIOUS: Makefile
+Makefile: $(srcdir)/Makefile.in $(top_builddir)/config.status
+	@case '$?' in \
+	  *config.status*) \
+	    cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh;; \
+	  *) \
+	    echo ' cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe)'; \
+	    cd $(top_builddir) && $(SHELL) ./config.status $(subdir)/$@ $(am__depfiles_maybe);; \
+	esac;
+
+$(top_builddir)/config.status: $(top_srcdir)/configure $(CONFIG_STATUS_DEPENDENCIES)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+
+$(top_srcdir)/configure: @MAINTAINER_MODE_TRUE@ $(am__configure_deps)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(ACLOCAL_M4): @MAINTAINER_MODE_TRUE@ $(am__aclocal_m4_deps)
+	cd $(top_builddir) && $(MAKE) $(AM_MAKEFLAGS) am--refresh
+$(am__aclocal_m4_deps):
+
+install-pluginLTLIBRARIES: $(plugin_LTLIBRARIES)
+	@$(NORMAL_INSTALL)
+	@list='$(plugin_LTLIBRARIES)'; test -n "$(plugindir)" || list=; \
+	list2=; for p in $$list; do \
+	  if test -f $$p; then \
+	    list2="$$list2 $$p"; \
+	  else :; fi; \
+	done; \
+	test -z "$$list2" || { \
+	  echo " $(MKDIR_P) '$(DESTDIR)$(plugindir)'"; \
+	  $(MKDIR_P) "$(DESTDIR)$(plugindir)" || exit 1; \
+	  echo " $(LIBTOOL) $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=install $(INSTALL) $(INSTALL_STRIP_FLAG) $$list2 '$(DESTDIR)$(plugindir)'"; \
+	  $(LIBTOOL) $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=install $(INSTALL) $(INSTALL_STRIP_FLAG) $$list2 "$(DESTDIR)$(plugindir)"; \
+	}
+
+uninstall-pluginLTLIBRARIES:
+	@$(NORMAL_UNINSTALL)
+	@list='$(plugin_LTLIBRARIES)'; test -n "$(plugindir)" || list=; \
+	for p in $$list; do \
+	  $(am__strip_dir) \
+	  echo " $(LIBTOOL) $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=uninstall rm -f '$(DESTDIR)$(plugindir)/$$f'"; \
+	  $(LIBTOOL) $(AM_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=uninstall rm -f "$(DESTDIR)$(plugindir)/$$f"; \
+	done
+
+clean-pluginLTLIBRARIES:
+	-test -z "$(plugin_LTLIBRARIES)" || rm -f $(plugin_LTLIBRARIES)
+	@list='$(plugin_LTLIBRARIES)'; \
+	locs=`for p in $$list; do echo $$p; done | \
+	      sed 's|^[^/]*$$|.|; s|/[^/]*$$||; s|$$|/so_locations|' | \
+	      sort -u`; \
+	test -z "$$locs" || { \
+	  echo rm -f $${locs}; \
+	  rm -f $${locs}; \
+	}
+
+libgstjack.la: $(libgstjack_la_OBJECTS) $(libgstjack_la_DEPENDENCIES) $(EXTRA_libgstjack_la_DEPENDENCIES) 
+	$(AM_V_CCLD)$(libgstjack_la_LINK) -rpath $(plugindir) $(libgstjack_la_OBJECTS) $(libgstjack_la_LIBADD) $(LIBS)
+
+mostlyclean-compile:
+	-rm -f *.$(OBJEXT)
+
+distclean-compile:
+	-rm -f *.tab.c
+
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstjack_la-gstjack.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstjack_la-gstjackaudioclient.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstjack_la-gstjackaudiosink.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstjack_la-gstjackaudiosrc.Plo@am__quote@
+@AMDEP_TRUE@@am__include@ @am__quote@./$(DEPDIR)/libgstjack_la-gstjackutil.Plo@am__quote@
+
+.c.o:
+@am__fastdepCC_TRUE@	$(AM_V_CC)depbase=`echo $@ | sed 's|[^/]*$$|$(DEPDIR)/&|;s|\.o$$||'`;\
+@am__fastdepCC_TRUE@	$(COMPILE) -MT $@ -MD -MP -MF $$depbase.Tpo -c -o $@ $< &&\
+@am__fastdepCC_TRUE@	$(am__mv) $$depbase.Tpo $$depbase.Po
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	$(AM_V_CC)source='$<' object='$@' libtool=no @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@	$(AM_V_CC@am__nodep@)$(COMPILE) -c -o $@ $<
+
+.c.obj:
+@am__fastdepCC_TRUE@	$(AM_V_CC)depbase=`echo $@ | sed 's|[^/]*$$|$(DEPDIR)/&|;s|\.obj$$||'`;\
+@am__fastdepCC_TRUE@	$(COMPILE) -MT $@ -MD -MP -MF $$depbase.Tpo -c -o $@ `$(CYGPATH_W) '$<'` &&\
+@am__fastdepCC_TRUE@	$(am__mv) $$depbase.Tpo $$depbase.Po
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	$(AM_V_CC)source='$<' object='$@' libtool=no @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@	$(AM_V_CC@am__nodep@)$(COMPILE) -c -o $@ `$(CYGPATH_W) '$<'`
+
+.c.lo:
+@am__fastdepCC_TRUE@	$(AM_V_CC)depbase=`echo $@ | sed 's|[^/]*$$|$(DEPDIR)/&|;s|\.lo$$||'`;\
+@am__fastdepCC_TRUE@	$(LTCOMPILE) -MT $@ -MD -MP -MF $$depbase.Tpo -c -o $@ $< &&\
+@am__fastdepCC_TRUE@	$(am__mv) $$depbase.Tpo $$depbase.Plo
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	$(AM_V_CC)source='$<' object='$@' libtool=yes @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@	$(AM_V_CC@am__nodep@)$(LTCOMPILE) -c -o $@ $<
+
+libgstjack_la-gstjackutil.lo: gstjackutil.c
+@am__fastdepCC_TRUE@	$(AM_V_CC)$(LIBTOOL) $(AM_V_lt) --tag=CC $(libgstjack_la_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libgstjack_la_CFLAGS) $(CFLAGS) -MT libgstjack_la-gstjackutil.lo -MD -MP -MF $(DEPDIR)/libgstjack_la-gstjackutil.Tpo -c -o libgstjack_la-gstjackutil.lo `test -f 'gstjackutil.c' || echo '$(srcdir)/'`gstjackutil.c
+@am__fastdepCC_TRUE@	$(AM_V_at)$(am__mv) $(DEPDIR)/libgstjack_la-gstjackutil.Tpo $(DEPDIR)/libgstjack_la-gstjackutil.Plo
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	$(AM_V_CC)source='gstjackutil.c' object='libgstjack_la-gstjackutil.lo' libtool=yes @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@	$(AM_V_CC@am__nodep@)$(LIBTOOL) $(AM_V_lt) --tag=CC $(libgstjack_la_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libgstjack_la_CFLAGS) $(CFLAGS) -c -o libgstjack_la-gstjackutil.lo `test -f 'gstjackutil.c' || echo '$(srcdir)/'`gstjackutil.c
+
+libgstjack_la-gstjack.lo: gstjack.c
+@am__fastdepCC_TRUE@	$(AM_V_CC)$(LIBTOOL) $(AM_V_lt) --tag=CC $(libgstjack_la_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libgstjack_la_CFLAGS) $(CFLAGS) -MT libgstjack_la-gstjack.lo -MD -MP -MF $(DEPDIR)/libgstjack_la-gstjack.Tpo -c -o libgstjack_la-gstjack.lo `test -f 'gstjack.c' || echo '$(srcdir)/'`gstjack.c
+@am__fastdepCC_TRUE@	$(AM_V_at)$(am__mv) $(DEPDIR)/libgstjack_la-gstjack.Tpo $(DEPDIR)/libgstjack_la-gstjack.Plo
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	$(AM_V_CC)source='gstjack.c' object='libgstjack_la-gstjack.lo' libtool=yes @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@	$(AM_V_CC@am__nodep@)$(LIBTOOL) $(AM_V_lt) --tag=CC $(libgstjack_la_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libgstjack_la_CFLAGS) $(CFLAGS) -c -o libgstjack_la-gstjack.lo `test -f 'gstjack.c' || echo '$(srcdir)/'`gstjack.c
+
+libgstjack_la-gstjackaudiosrc.lo: gstjackaudiosrc.c
+@am__fastdepCC_TRUE@	$(AM_V_CC)$(LIBTOOL) $(AM_V_lt) --tag=CC $(libgstjack_la_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libgstjack_la_CFLAGS) $(CFLAGS) -MT libgstjack_la-gstjackaudiosrc.lo -MD -MP -MF $(DEPDIR)/libgstjack_la-gstjackaudiosrc.Tpo -c -o libgstjack_la-gstjackaudiosrc.lo `test -f 'gstjackaudiosrc.c' || echo '$(srcdir)/'`gstjackaudiosrc.c
+@am__fastdepCC_TRUE@	$(AM_V_at)$(am__mv) $(DEPDIR)/libgstjack_la-gstjackaudiosrc.Tpo $(DEPDIR)/libgstjack_la-gstjackaudiosrc.Plo
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	$(AM_V_CC)source='gstjackaudiosrc.c' object='libgstjack_la-gstjackaudiosrc.lo' libtool=yes @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@	$(AM_V_CC@am__nodep@)$(LIBTOOL) $(AM_V_lt) --tag=CC $(libgstjack_la_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libgstjack_la_CFLAGS) $(CFLAGS) -c -o libgstjack_la-gstjackaudiosrc.lo `test -f 'gstjackaudiosrc.c' || echo '$(srcdir)/'`gstjackaudiosrc.c
+
+libgstjack_la-gstjackaudiosink.lo: gstjackaudiosink.c
+@am__fastdepCC_TRUE@	$(AM_V_CC)$(LIBTOOL) $(AM_V_lt) --tag=CC $(libgstjack_la_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libgstjack_la_CFLAGS) $(CFLAGS) -MT libgstjack_la-gstjackaudiosink.lo -MD -MP -MF $(DEPDIR)/libgstjack_la-gstjackaudiosink.Tpo -c -o libgstjack_la-gstjackaudiosink.lo `test -f 'gstjackaudiosink.c' || echo '$(srcdir)/'`gstjackaudiosink.c
+@am__fastdepCC_TRUE@	$(AM_V_at)$(am__mv) $(DEPDIR)/libgstjack_la-gstjackaudiosink.Tpo $(DEPDIR)/libgstjack_la-gstjackaudiosink.Plo
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	$(AM_V_CC)source='gstjackaudiosink.c' object='libgstjack_la-gstjackaudiosink.lo' libtool=yes @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@	$(AM_V_CC@am__nodep@)$(LIBTOOL) $(AM_V_lt) --tag=CC $(libgstjack_la_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libgstjack_la_CFLAGS) $(CFLAGS) -c -o libgstjack_la-gstjackaudiosink.lo `test -f 'gstjackaudiosink.c' || echo '$(srcdir)/'`gstjackaudiosink.c
+
+libgstjack_la-gstjackaudioclient.lo: gstjackaudioclient.c
+@am__fastdepCC_TRUE@	$(AM_V_CC)$(LIBTOOL) $(AM_V_lt) --tag=CC $(libgstjack_la_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libgstjack_la_CFLAGS) $(CFLAGS) -MT libgstjack_la-gstjackaudioclient.lo -MD -MP -MF $(DEPDIR)/libgstjack_la-gstjackaudioclient.Tpo -c -o libgstjack_la-gstjackaudioclient.lo `test -f 'gstjackaudioclient.c' || echo '$(srcdir)/'`gstjackaudioclient.c
+@am__fastdepCC_TRUE@	$(AM_V_at)$(am__mv) $(DEPDIR)/libgstjack_la-gstjackaudioclient.Tpo $(DEPDIR)/libgstjack_la-gstjackaudioclient.Plo
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	$(AM_V_CC)source='gstjackaudioclient.c' object='libgstjack_la-gstjackaudioclient.lo' libtool=yes @AMDEPBACKSLASH@
+@AMDEP_TRUE@@am__fastdepCC_FALSE@	DEPDIR=$(DEPDIR) $(CCDEPMODE) $(depcomp) @AMDEPBACKSLASH@
+@am__fastdepCC_FALSE@	$(AM_V_CC@am__nodep@)$(LIBTOOL) $(AM_V_lt) --tag=CC $(libgstjack_la_LIBTOOLFLAGS) $(LIBTOOLFLAGS) --mode=compile $(CC) $(DEFS) $(DEFAULT_INCLUDES) $(INCLUDES) $(AM_CPPFLAGS) $(CPPFLAGS) $(libgstjack_la_CFLAGS) $(CFLAGS) -c -o libgstjack_la-gstjackaudioclient.lo `test -f 'gstjackaudioclient.c' || echo '$(srcdir)/'`gstjackaudioclient.c
+
+mostlyclean-libtool:
+	-rm -f *.lo
+
+clean-libtool:
+	-rm -rf .libs _libs
+
+ID: $(am__tagged_files)
+	$(am__define_uniq_tagged_files); mkid -fID $$unique
+tags: tags-am
+TAGS: tags
+
+tags-am: $(TAGS_DEPENDENCIES) $(am__tagged_files)
+	set x; \
+	here=`pwd`; \
+	$(am__define_uniq_tagged_files); \
+	shift; \
+	if test -z "$(ETAGS_ARGS)$$*$$unique"; then :; else \
+	  test -n "$$unique" || unique=$$empty_fix; \
+	  if test $$# -gt 0; then \
+	    $(ETAGS) $(ETAGSFLAGS) $(AM_ETAGSFLAGS) $(ETAGS_ARGS) \
+	      "$$@" $$unique; \
+	  else \
+	    $(ETAGS) $(ETAGSFLAGS) $(AM_ETAGSFLAGS) $(ETAGS_ARGS) \
+	      $$unique; \
+	  fi; \
+	fi
+ctags: ctags-am
+
+CTAGS: ctags
+ctags-am: $(TAGS_DEPENDENCIES) $(am__tagged_files)
+	$(am__define_uniq_tagged_files); \
+	test -z "$(CTAGS_ARGS)$$unique" \
+	  || $(CTAGS) $(CTAGSFLAGS) $(AM_CTAGSFLAGS) $(CTAGS_ARGS) \
+	     $$unique
+
+GTAGS:
+	here=`$(am__cd) $(top_builddir) && pwd` \
+	  && $(am__cd) $(top_srcdir) \
+	  && gtags -i $(GTAGS_ARGS) "$$here"
+cscopelist: cscopelist-am
+
+cscopelist-am: $(am__tagged_files)
+	list='$(am__tagged_files)'; \
+	case "$(srcdir)" in \
+	  [\\/]* | ?:[\\/]*) sdir="$(srcdir)" ;; \
+	  *) sdir=$(subdir)/$(srcdir) ;; \
+	esac; \
+	for i in $$list; do \
+	  if test -f "$$i"; then \
+	    echo "$(subdir)/$$i"; \
+	  else \
+	    echo "$$sdir/$$i"; \
+	  fi; \
+	done >> $(top_builddir)/cscope.files
+
+distclean-tags:
+	-rm -f TAGS ID GTAGS GRTAGS GSYMS GPATH tags
+
+distdir: $(DISTFILES)
+	@srcdirstrip=`echo "$(srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	topsrcdirstrip=`echo "$(top_srcdir)" | sed 's/[].[^$$\\*]/\\\\&/g'`; \
+	list='$(DISTFILES)'; \
+	  dist_files=`for file in $$list; do echo $$file; done | \
+	  sed -e "s|^$$srcdirstrip/||;t" \
+	      -e "s|^$$topsrcdirstrip/|$(top_builddir)/|;t"`; \
+	case $$dist_files in \
+	  */*) $(MKDIR_P) `echo "$$dist_files" | \
+			   sed '/\//!d;s|^|$(distdir)/|;s,/[^/]*$$,,' | \
+			   sort -u` ;; \
+	esac; \
+	for file in $$dist_files; do \
+	  if test -f $$file || test -d $$file; then d=.; else d=$(srcdir); fi; \
+	  if test -d $$d/$$file; then \
+	    dir=`echo "/$$file" | sed -e 's,/[^/]*$$,,'`; \
+	    if test -d "$(distdir)/$$file"; then \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    if test -d $(srcdir)/$$file && test $$d != $(srcdir); then \
+	      cp -fpR $(srcdir)/$$file "$(distdir)$$dir" || exit 1; \
+	      find "$(distdir)/$$file" -type d ! -perm -700 -exec chmod u+rwx {} \;; \
+	    fi; \
+	    cp -fpR $$d/$$file "$(distdir)$$dir" || exit 1; \
+	  else \
+	    test -f "$(distdir)/$$file" \
+	    || cp -p $$d/$$file "$(distdir)/$$file" \
+	    || exit 1; \
+	  fi; \
+	done
+check-am: all-am
+check: check-am
+all-am: Makefile $(LTLIBRARIES) $(HEADERS)
+installdirs:
+	for dir in "$(DESTDIR)$(plugindir)"; do \
+	  test -z "$$dir" || $(MKDIR_P) "$$dir"; \
+	done
+install: install-am
+install-exec: install-exec-am
+install-data: install-data-am
+uninstall: uninstall-am
+
+install-am: all-am
+	@$(MAKE) $(AM_MAKEFLAGS) install-exec-am install-data-am
+
+installcheck: installcheck-am
+install-strip:
+	if test -z '$(STRIP)'; then \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	      install; \
+	else \
+	  $(MAKE) $(AM_MAKEFLAGS) INSTALL_PROGRAM="$(INSTALL_STRIP_PROGRAM)" \
+	    install_sh_PROGRAM="$(INSTALL_STRIP_PROGRAM)" INSTALL_STRIP_FLAG=-s \
+	    "INSTALL_PROGRAM_ENV=STRIPPROG='$(STRIP)'" install; \
+	fi
+mostlyclean-generic:
+
+clean-generic:
+
+distclean-generic:
+	-test -z "$(CONFIG_CLEAN_FILES)" || rm -f $(CONFIG_CLEAN_FILES)
+	-test . = "$(srcdir)" || test -z "$(CONFIG_CLEAN_VPATH_FILES)" || rm -f $(CONFIG_CLEAN_VPATH_FILES)
+
+maintainer-clean-generic:
+	@echo "This command is intended for maintainers to use"
+	@echo "it deletes files that may require special tools to rebuild."
+clean: clean-am
+
+clean-am: clean-generic clean-libtool clean-pluginLTLIBRARIES \
+	mostlyclean-am
+
+distclean: distclean-am
+	-rm -rf ./$(DEPDIR)
+	-rm -f Makefile
+distclean-am: clean-am distclean-compile distclean-generic \
+	distclean-tags
+
+dvi: dvi-am
+
+dvi-am:
+
+html: html-am
+
+html-am:
+
+info: info-am
+
+info-am:
+
+install-data-am: install-pluginLTLIBRARIES
+
+install-dvi: install-dvi-am
+
+install-dvi-am:
+
+install-exec-am:
+
+install-html: install-html-am
+
+install-html-am:
+
+install-info: install-info-am
+
+install-info-am:
+
+install-man:
+
+install-pdf: install-pdf-am
+
+install-pdf-am:
+
+install-ps: install-ps-am
+
+install-ps-am:
+
+installcheck-am:
+
+maintainer-clean: maintainer-clean-am
+	-rm -rf ./$(DEPDIR)
+	-rm -f Makefile
+maintainer-clean-am: distclean-am maintainer-clean-generic
+
+mostlyclean: mostlyclean-am
+
+mostlyclean-am: mostlyclean-compile mostlyclean-generic \
+	mostlyclean-libtool
+
+pdf: pdf-am
+
+pdf-am:
+
+ps: ps-am
+
+ps-am:
+
+uninstall-am: uninstall-pluginLTLIBRARIES
+
+.MAKE: install-am install-strip
+
+.PHONY: CTAGS GTAGS TAGS all all-am check check-am clean clean-generic \
+	clean-libtool clean-pluginLTLIBRARIES cscopelist-am ctags \
+	ctags-am distclean distclean-compile distclean-generic \
+	distclean-libtool distclean-tags distdir dvi dvi-am html \
+	html-am info info-am install install-am install-data \
+	install-data-am install-dvi install-dvi-am install-exec \
+	install-exec-am install-html install-html-am install-info \
+	install-info-am install-man install-pdf install-pdf-am \
+	install-pluginLTLIBRARIES install-ps install-ps-am \
+	install-strip installcheck installcheck-am installdirs \
+	maintainer-clean maintainer-clean-generic mostlyclean \
+	mostlyclean-compile mostlyclean-generic mostlyclean-libtool \
+	pdf pdf-am ps ps-am tags tags-am uninstall uninstall-am \
+	uninstall-pluginLTLIBRARIES
+
+
+# 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:
diff --git a/ext/jack/gstjack.c b/ext/jack/gstjack.c
new file mode 100644
index 0000000..ca98dc4
--- /dev/null
+++ b/ext/jack/gstjack.c
@@ -0,0 +1,117 @@
+/* GStreamer Jack plugins
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include "gstjack.h"
+#include "gstjackaudiosrc.h"
+#include "gstjackaudiosink.h"
+
+GType
+gst_jack_connect_get_type (void)
+{
+  static volatile gsize jack_connect_type = 0;
+
+  if (g_once_init_enter (&jack_connect_type)) {
+    static const GEnumValue jack_connect_enums[] = {
+      {GST_JACK_CONNECT_NONE,
+          "Don't automatically connect ports to physical ports", "none"},
+      {GST_JACK_CONNECT_AUTO,
+          "Automatically connect ports to physical ports", "auto"},
+      {GST_JACK_CONNECT_AUTO_FORCED,
+            "Automatically connect ports to as many physical ports as possible",
+          "auto-forced"},
+      {0, NULL, NULL},
+    };
+    GType tmp = g_enum_register_static ("GstJackConnect", jack_connect_enums);
+    g_once_init_leave (&jack_connect_type, tmp);
+  }
+  return (GType) jack_connect_type;
+}
+
+GType
+gst_jack_transport_get_type (void)
+{
+  static volatile gsize type = 0;
+
+  if (g_once_init_enter (&type)) {
+    static const GFlagsValue flag_values[] = {
+      {GST_JACK_TRANSPORT_MASTER,
+          "Start and stop transport with state changes", "master"},
+      {GST_JACK_TRANSPORT_SLAVE,
+          "Follow transport state changes", "slave"},
+      {0, NULL, NULL},
+    };
+    GType tmp = g_flags_register_static ("GstJackTransport", flag_values);
+    g_once_init_leave (&type, tmp);
+  }
+  return (GType) type;
+}
+
+
+static gpointer
+gst_jack_client_copy (gpointer jclient)
+{
+  return jclient;
+}
+
+
+static void
+gst_jack_client_free (gpointer jclient)
+{
+  return;
+}
+
+
+GType
+gst_jack_client_get_type (void)
+{
+  static volatile gsize jack_client_type = 0;
+
+  if (g_once_init_enter (&jack_client_type)) {
+    /* hackish, but makes it show up nicely in gst-inspect */
+    GType tmp = g_boxed_type_register_static ("JackClient",
+        (GBoxedCopyFunc) gst_jack_client_copy,
+        (GBoxedFreeFunc) gst_jack_client_free);
+    g_once_init_leave (&jack_client_type, tmp);
+  }
+
+  return (GType) jack_client_type;
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  if (!gst_element_register (plugin, "jackaudiosrc", GST_RANK_PRIMARY,
+          GST_TYPE_JACK_AUDIO_SRC))
+    return FALSE;
+  if (!gst_element_register (plugin, "jackaudiosink", GST_RANK_PRIMARY,
+          GST_TYPE_JACK_AUDIO_SINK))
+    return FALSE;
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
+    GST_VERSION_MINOR,
+    jack,
+    "JACK audio elements",
+    plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)
diff --git a/ext/jack/gstjack.h b/ext/jack/gstjack.h
new file mode 100644
index 0000000..15b040e
--- /dev/null
+++ b/ext/jack/gstjack.h
@@ -0,0 +1,79 @@
+/* GStreamer
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ *
+ * gstjack.h:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef _GST_JACK_H_
+#define _GST_JACK_H_
+
+#include <jack/jack.h>
+#include <gst/audio/audio.h>
+
+/**
+ * GstJackConnect:
+ * @GST_JACK_CONNECT_NONE: Don't automatically connect to physical ports.
+ *     In this mode, the element will accept any number of input channels and will
+ *     create (but not connect) an output port for each channel.
+ * @GST_JACK_CONNECT_AUTO: In this mode, the element will try to connect each
+ *     output port to a random physical jack input pin. The sink will
+ *     expose the number of physical channels on its pad caps.
+ * @GST_JACK_CONNECT_AUTO_FORCED: In this mode, the element will try to connect each
+ *     output port to a random physical jack input pin. The  element will accept any number
+ *     of input channels.
+ *
+ * Specify how the output ports will be connected.
+ */
+typedef enum {
+  GST_JACK_CONNECT_NONE,
+  GST_JACK_CONNECT_AUTO,
+  GST_JACK_CONNECT_AUTO_FORCED
+} GstJackConnect;
+
+/**
+ * GstJackTransport:
+ * @GST_JACK_TRANSPORT_AUTONOMOUS: no transport support
+ * @GST_JACK_TRANSPORT_MASTER: start and stop transport with state-changes
+ * @GST_JACK_TRANSPORT_SLAVE: follow transport state changes
+ *
+ * The jack transport state allow to sync multiple clients. This enum defines a
+ * client behaviour regarding to the transport mechanism.
+ */
+typedef enum {
+  GST_JACK_TRANSPORT_AUTONOMOUS = 0,
+  GST_JACK_TRANSPORT_MASTER = (1 << 0),
+  GST_JACK_TRANSPORT_SLAVE = (1 << 1),
+} GstJackTransport;
+
+typedef jack_default_audio_sample_t sample_t;
+
+#define GST_TYPE_JACK_CONNECT   (gst_jack_connect_get_type ())
+#define GST_TYPE_JACK_TRANSPORT (gst_jack_transport_get_type ())
+#define GST_TYPE_JACK_CLIENT    (gst_jack_client_get_type ())
+
+#if G_BYTE_ORDER == G_LITTLE_ENDIAN
+#define GST_JACK_FORMAT_STR "F32LE"
+#else
+#define GST_JACK_FORMAT_STR "F32BE"
+#endif
+
+GType gst_jack_client_get_type(void);
+GType gst_jack_connect_get_type(void);
+GType gst_jack_transport_get_type(void);
+
+#endif  // _GST_JACK_H_
diff --git a/ext/jack/gstjackaudioclient.c b/ext/jack/gstjackaudioclient.c
new file mode 100755
index 0000000..0f06d10
--- /dev/null
+++ b/ext/jack/gstjackaudioclient.c
@@ -0,0 +1,629 @@
+/* GStreamer
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ *
+ * gstjackaudioclient.c: jack audio client implementation
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include <string.h>
+
+#include "gstjackaudioclient.h"
+#include "gstjack.h"
+
+#include <gst/glib-compat-private.h>
+
+GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_client_debug);
+#define GST_CAT_DEFAULT gst_jack_audio_client_debug
+
+static void
+jack_log_error (const gchar * msg)
+{
+  GST_ERROR ("%s", msg);
+}
+
+static void
+jack_info_error (const gchar * msg)
+{
+  GST_INFO ("%s", msg);
+}
+
+void
+gst_jack_audio_client_init (void)
+{
+  GST_DEBUG_CATEGORY_INIT (gst_jack_audio_client_debug, "jackclient", 0,
+      "jackclient helpers");
+
+  jack_set_error_function (jack_log_error);
+  jack_set_info_function (jack_info_error);
+}
+
+/* a list of global connections indexed by id and server. */
+G_LOCK_DEFINE_STATIC (connections_lock);
+static GList *connections;
+
+/* the connection to a server  */
+typedef struct
+{
+  gint refcount;
+  GMutex lock;
+  GCond flush_cond;
+
+  /* id/server pair and the connection */
+  gchar *id;
+  gchar *server;
+  jack_client_t *client;
+
+  /* lists of GstJackAudioClients */
+  gint n_clients;
+  GList *src_clients;
+  GList *sink_clients;
+
+  /* transport state handling */
+  gint cur_ts;
+  GstState transport_state;
+} GstJackAudioConnection;
+
+/* an object sharing a jack_client_t connection. */
+struct _GstJackAudioClient
+{
+  GstJackAudioConnection *conn;
+
+  GstJackClientType type;
+  gboolean active;
+  gboolean deactivate;
+
+  JackShutdownCallback shutdown;
+  JackProcessCallback process;
+  JackBufferSizeCallback buffer_size;
+  JackSampleRateCallback sample_rate;
+  gpointer user_data;
+};
+
+typedef struct
+{
+  jack_nframes_t nframes;
+  gpointer user_data;
+} JackCB;
+
+static gboolean
+jack_handle_transport_change (GstJackAudioClient * client, GstState state)
+{
+  GstObject *obj = GST_OBJECT_PARENT (client->user_data);
+  guint mode;
+
+  g_object_get (obj, "transport", &mode, NULL);
+  if ((mode & GST_JACK_TRANSPORT_SLAVE) && (GST_STATE (obj) != state)) {
+    GST_INFO_OBJECT (obj, "requesting state change: %s",
+        gst_element_state_get_name (state));
+    gst_element_post_message (GST_ELEMENT (obj),
+        gst_message_new_request_state (obj, state));
+    return TRUE;
+  }
+  return FALSE;
+}
+
+static int
+jack_process_cb (jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
+  GList *walk;
+  int res = 0;
+  jack_transport_state_t ts = jack_transport_query (conn->client, NULL);
+
+  if (ts != conn->cur_ts) {
+    conn->cur_ts = ts;
+    switch (ts) {
+      case JackTransportStopped:
+        GST_DEBUG ("transport state is 'stopped'");
+        conn->transport_state = GST_STATE_PAUSED;
+        break;
+      case JackTransportStarting:
+        GST_DEBUG ("transport state is 'starting'");
+        conn->transport_state = GST_STATE_READY;
+        break;
+      case JackTransportRolling:
+        GST_DEBUG ("transport state is 'rolling'");
+        conn->transport_state = GST_STATE_PLAYING;
+        break;
+      default:
+        break;
+    }
+    GST_DEBUG ("num of clients: src=%d, sink=%d",
+        g_list_length (conn->src_clients), g_list_length (conn->sink_clients));
+  }
+
+  g_mutex_lock (&conn->lock);
+  /* call sources first, then sinks. Sources will either push data into the
+   * ringbuffer of the sinks, which will then pull the data out of it, or
+   * sinks will pull the data from the sources. */
+  for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
+    GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
+
+    /* only call active clients */
+    if ((client->active || client->deactivate) && client->process) {
+      res = client->process (nframes, client->user_data);
+      if (client->deactivate) {
+        client->deactivate = FALSE;
+        g_cond_signal (&conn->flush_cond);
+      }
+    }
+  }
+  for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
+    GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
+
+    /* only call active clients */
+    if ((client->active || client->deactivate) && client->process) {
+      res = client->process (nframes, client->user_data);
+      if (client->deactivate) {
+        client->deactivate = FALSE;
+        g_cond_signal (&conn->flush_cond);
+      }
+    }
+  }
+
+  /* handle transport state requisition, do sinks first, stop after the first
+   * element that handled it */
+  if (conn->transport_state != GST_STATE_VOID_PENDING) {
+    for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
+      if (jack_handle_transport_change ((GstJackAudioClient *) walk->data,
+              conn->transport_state)) {
+        conn->transport_state = GST_STATE_VOID_PENDING;
+        break;
+      }
+    }
+  }
+  if (conn->transport_state != GST_STATE_VOID_PENDING) {
+    for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
+      if (jack_handle_transport_change ((GstJackAudioClient *) walk->data,
+              conn->transport_state)) {
+        conn->transport_state = GST_STATE_VOID_PENDING;
+        break;
+      }
+    }
+  }
+  g_mutex_unlock (&conn->lock);
+
+  return res;
+}
+
+/* we error out */
+static int
+jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
+{
+  return 0;
+}
+
+/* we error out */
+static int
+jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
+{
+  return 0;
+}
+
+static void
+jack_shutdown_cb (void *arg)
+{
+  GstJackAudioConnection *conn = (GstJackAudioConnection *) arg;
+  GList *walk;
+
+  GST_DEBUG ("disconnect client %s from server %s", conn->id,
+      GST_STR_NULL (conn->server));
+
+  g_mutex_lock (&conn->lock);
+  for (walk = conn->src_clients; walk; walk = g_list_next (walk)) {
+    GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
+
+    if (client->shutdown)
+      client->shutdown (client->user_data);
+  }
+  for (walk = conn->sink_clients; walk; walk = g_list_next (walk)) {
+    GstJackAudioClient *client = (GstJackAudioClient *) walk->data;
+
+    if (client->shutdown)
+      client->shutdown (client->user_data);
+  }
+  g_mutex_unlock (&conn->lock);
+}
+
+typedef struct
+{
+  const gchar *id;
+  const gchar *server;
+} FindData;
+
+static gint
+connection_find (GstJackAudioConnection * conn, FindData * data)
+{
+  /* id's must match */
+  if (strcmp (conn->id, data->id))
+    return 1;
+
+  /* both the same or NULL */
+  if (conn->server == data->server)
+    return 0;
+
+  /* we cannot compare NULL */
+  if (conn->server == NULL || data->server == NULL)
+    return 1;
+
+  if (strcmp (conn->server, data->server))
+    return 1;
+
+  return 0;
+}
+
+/* make a connection with @id and @server. Returns NULL on failure with the
+ * status set. */
+static GstJackAudioConnection *
+gst_jack_audio_make_connection (const gchar * id, const gchar * server,
+    jack_client_t * jclient, jack_status_t * status)
+{
+  GstJackAudioConnection *conn;
+  jack_options_t options;
+  gint res;
+
+  *status = 0;
+
+  GST_DEBUG ("new client %s, connecting to server %s", id,
+      GST_STR_NULL (server));
+
+  /* never start a server */
+  options = JackNoStartServer;
+  /* if we have a servername, use it */
+  if (server != NULL)
+    options |= JackServerName;
+  /* open the client */
+  if (jclient == NULL)
+    jclient = jack_client_open (id, options, status, server);
+  if (jclient == NULL)
+    goto could_not_open;
+
+  /* now create object */
+  conn = g_new (GstJackAudioConnection, 1);
+  conn->refcount = 1;
+  g_mutex_init (&conn->lock);
+  g_cond_init (&conn->flush_cond);
+  conn->id = g_strdup (id);
+  conn->server = g_strdup (server);
+  conn->client = jclient;
+  conn->n_clients = 0;
+  conn->src_clients = NULL;
+  conn->sink_clients = NULL;
+  conn->cur_ts = -1;
+  conn->transport_state = GST_STATE_VOID_PENDING;
+
+  /* set our callbacks  */
+  jack_set_process_callback (jclient, jack_process_cb, conn);
+  /* these callbacks cause us to error */
+  jack_set_buffer_size_callback (jclient, jack_buffer_size_cb, conn);
+  jack_set_sample_rate_callback (jclient, jack_sample_rate_cb, conn);
+  jack_on_shutdown (jclient, jack_shutdown_cb, conn);
+
+  /* all callbacks are set, activate the client */
+  GST_INFO ("activate jack_client %p", jclient);
+  if ((res = jack_activate (jclient)))
+    goto could_not_activate;
+
+  GST_DEBUG ("opened connection %p", conn);
+
+  return conn;
+
+  /* ERRORS */
+could_not_open:
+  {
+    GST_DEBUG ("failed to open jack client, %d", *status);
+    return NULL;
+  }
+could_not_activate:
+  {
+    GST_ERROR ("Could not activate client (%d)", res);
+    *status = JackFailure;
+    g_mutex_clear (&conn->lock);
+    g_free (conn->id);
+    g_free (conn->server);
+    g_free (conn);
+    return NULL;
+  }
+}
+
+static GstJackAudioConnection *
+gst_jack_audio_get_connection (const gchar * id, const gchar * server,
+    jack_client_t * jclient, jack_status_t * status)
+{
+  GstJackAudioConnection *conn;
+  GList *found;
+  FindData data;
+
+  GST_DEBUG ("getting connection for id %s, server %s", id,
+      GST_STR_NULL (server));
+
+  data.id = id;
+  data.server = server;
+
+  G_LOCK (connections_lock);
+  found =
+      g_list_find_custom (connections, &data, (GCompareFunc) connection_find);
+  if (found != NULL && jclient != NULL) {
+    /* we found it, increase refcount and return it */
+    conn = (GstJackAudioConnection *) found->data;
+    conn->refcount++;
+
+    GST_DEBUG ("found connection %p", conn);
+  } else {
+    /* make new connection */
+    conn = gst_jack_audio_make_connection (id, server, jclient, status);
+    if (conn != NULL) {
+      GST_DEBUG ("created connection %p", conn);
+      /* add to list on success */
+      connections = g_list_prepend (connections, conn);
+    } else {
+      GST_WARNING ("could not create connection");
+    }
+  }
+  G_UNLOCK (connections_lock);
+
+  return conn;
+}
+
+static void
+gst_jack_audio_unref_connection (GstJackAudioConnection * conn)
+{
+  gint res;
+  gboolean zero;
+
+  GST_DEBUG ("unref connection %p refcnt %d", conn, conn->refcount);
+
+  G_LOCK (connections_lock);
+  conn->refcount--;
+  if ((zero = (conn->refcount == 0))) {
+    GST_DEBUG ("closing connection %p", conn);
+    /* remove from list, we can release the mutex after removing the connection
+     * from the list because after that, nobody can access the connection anymore. */
+    connections = g_list_remove (connections, conn);
+  }
+  G_UNLOCK (connections_lock);
+
+  /* if we are zero, close and cleanup the connection */
+  if (zero) {
+    /* don't use conn->lock here. two reasons:
+     *
+     *  1) its not necessary: jack_deactivate() will not return until the JACK thread
+     *      associated with this connection is cleaned up by a thread join, hence 
+     *      no more callbacks can occur or be in progress.
+     *
+     * 2) it would deadlock anyway, because jack_deactivate() will sleep
+     *      waiting for the JACK thread, and can thus cause deadlock in 
+     *      jack_process_cb()
+     */
+    GST_INFO ("deactivate jack_client %p", conn->client);
+    if ((res = jack_deactivate (conn->client))) {
+      /* we only warn, this means the server is probably shut down and the client
+       * is gone anyway. */
+      GST_WARNING ("Could not deactivate Jack client (%d)", res);
+    }
+    /* close connection */
+    if ((res = jack_client_close (conn->client))) {
+      /* we assume the client is gone. */
+      GST_WARNING ("close failed (%d)", res);
+    }
+
+    /* free resources */
+    g_mutex_clear (&conn->lock);
+    g_cond_clear (&conn->flush_cond);
+    g_free (conn->id);
+    g_free (conn->server);
+    g_free (conn);
+  }
+}
+
+static void
+gst_jack_audio_connection_add_client (GstJackAudioConnection * conn,
+    GstJackAudioClient * client)
+{
+  g_mutex_lock (&conn->lock);
+  switch (client->type) {
+    case GST_JACK_CLIENT_SOURCE:
+      conn->src_clients = g_list_append (conn->src_clients, client);
+      conn->n_clients++;
+      break;
+    case GST_JACK_CLIENT_SINK:
+      conn->sink_clients = g_list_append (conn->sink_clients, client);
+      conn->n_clients++;
+      break;
+    default:
+      g_warning ("trying to add unknown client type");
+      break;
+  }
+  g_mutex_unlock (&conn->lock);
+}
+
+static void
+gst_jack_audio_connection_remove_client (GstJackAudioConnection * conn,
+    GstJackAudioClient * client)
+{
+  g_mutex_lock (&conn->lock);
+  switch (client->type) {
+    case GST_JACK_CLIENT_SOURCE:
+      conn->src_clients = g_list_remove (conn->src_clients, client);
+      conn->n_clients--;
+      break;
+    case GST_JACK_CLIENT_SINK:
+      conn->sink_clients = g_list_remove (conn->sink_clients, client);
+      conn->n_clients--;
+      break;
+    default:
+      g_warning ("trying to remove unknown client type");
+      break;
+  }
+  g_mutex_unlock (&conn->lock);
+}
+
+/**
+ * gst_jack_audio_client_get:
+ * @id: the client id
+ * @server: the server to connect to or NULL for the default server
+ * @type: the client type
+ * @shutdown: a callback when the jack server shuts down
+ * @process: a callback when samples are available
+ * @buffer_size: a callback when the buffer_size changes
+ * @sample_rate: a callback when the sample_rate changes
+ * @user_data: user data passed to the callbacks
+ * @status: pointer to hold the jack status code in case of errors
+ *
+ * Get the jack client connection for @id and @server. Connections to the same
+ * @id and @server will receive the same physical Jack client connection and
+ * will therefore be scheduled in the same process callback.
+ * 
+ * Returns: a #GstJackAudioClient.
+ */
+GstJackAudioClient *
+gst_jack_audio_client_new (const gchar * id, const gchar * server,
+    jack_client_t * jclient, GstJackClientType type,
+    void (*shutdown) (void *arg), JackProcessCallback process,
+    JackBufferSizeCallback buffer_size, JackSampleRateCallback sample_rate,
+    gpointer user_data, jack_status_t * status)
+{
+  GstJackAudioClient *client;
+  GstJackAudioConnection *conn;
+
+  g_return_val_if_fail (id != NULL, NULL);
+  g_return_val_if_fail (status != NULL, NULL);
+
+  /* first get a connection for the id/server pair */
+  conn = gst_jack_audio_get_connection (id, server, jclient, status);
+  if (conn == NULL)
+    goto no_connection;
+
+  GST_INFO ("new client %s", id);
+
+  /* make new client using the connection */
+  client = g_new (GstJackAudioClient, 1);
+  client->active = client->deactivate = FALSE;
+  client->conn = conn;
+  client->type = type;
+  client->shutdown = shutdown;
+  client->process = process;
+  client->buffer_size = buffer_size;
+  client->sample_rate = sample_rate;
+  client->user_data = user_data;
+
+  /* add the client to the connection */
+  gst_jack_audio_connection_add_client (conn, client);
+
+  return client;
+
+  /* ERRORS */
+no_connection:
+  {
+    GST_DEBUG ("Could not get server connection (%d)", *status);
+    return NULL;
+  }
+}
+
+/**
+ * gst_jack_audio_client_free:
+ * @client: a #GstJackAudioClient
+ *
+ * Free the resources used by @client.
+ */
+void
+gst_jack_audio_client_free (GstJackAudioClient * client)
+{
+  GstJackAudioConnection *conn;
+
+  g_return_if_fail (client != NULL);
+
+  GST_INFO ("free client");
+
+  conn = client->conn;
+
+  /* remove from connection first so that it's not scheduled anymore after this
+   * call */
+  gst_jack_audio_connection_remove_client (conn, client);
+  gst_jack_audio_unref_connection (conn);
+
+  g_free (client);
+}
+
+/**
+ * gst_jack_audio_client_get_client:
+ * @client: a #GstJackAudioClient
+ *
+ * Get the jack audio client for @client. This function is used to perform
+ * operations on the jack server from this client.
+ *
+ * Returns: The jack audio client.
+ */
+jack_client_t *
+gst_jack_audio_client_get_client (GstJackAudioClient * client)
+{
+  g_return_val_if_fail (client != NULL, NULL);
+
+  /* no lock needed, the connection and the client does not change 
+   * once the client is created. */
+  return client->conn->client;
+}
+
+/**
+ * gst_jack_audio_client_set_active:
+ * @client: a #GstJackAudioClient
+ * @active: new mode for the client
+ *
+ * Activate or deactive @client. When a client is activated it will receive
+ * callbacks when data should be processed.
+ *
+ * Returns: 0 if all ok.
+ */
+gint
+gst_jack_audio_client_set_active (GstJackAudioClient * client, gboolean active)
+{
+  g_return_val_if_fail (client != NULL, -1);
+
+  /* make sure that we are not dispatching the client */
+  g_mutex_lock (&client->conn->lock);
+  if (client->active && !active) {
+    /* we need to process once more to flush the port */
+    client->deactivate = TRUE;
+
+    /* need to wait for process_cb run once more */
+    while (client->deactivate)
+      g_cond_wait (&client->conn->flush_cond, &client->conn->lock);
+  }
+  client->active = active;
+  g_mutex_unlock (&client->conn->lock);
+
+  return 0;
+}
+
+/**
+ * gst_jack_audio_client_get_transport_state:
+ * @client: a #GstJackAudioClient
+ *
+ * Check the current transport state. The client can use this to request a state
+ * change from the application.
+ *
+ * Returns: the state, %GST_STATE_VOID_PENDING for no change in the transport
+ * state
+ */
+GstState
+gst_jack_audio_client_get_transport_state (GstJackAudioClient * client)
+{
+  GstState state = client->conn->transport_state;
+
+  client->conn->transport_state = GST_STATE_VOID_PENDING;
+  return state;
+}
diff --git a/ext/jack/gstjackaudioclient.h b/ext/jack/gstjackaudioclient.h
new file mode 100644
index 0000000..5dcd70c
--- /dev/null
+++ b/ext/jack/gstjackaudioclient.h
@@ -0,0 +1,61 @@
+/* GStreamer
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ *
+ * gstjackaudioclient.h:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_JACK_AUDIO_CLIENT_H__
+#define __GST_JACK_AUDIO_CLIENT_H__
+
+#include <jack/jack.h>
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+typedef enum 
+{
+  GST_JACK_CLIENT_SOURCE,
+  GST_JACK_CLIENT_SINK
+} GstJackClientType;
+
+typedef struct _GstJackAudioClient GstJackAudioClient;
+
+void                  gst_jack_audio_client_init           (void);
+
+
+GstJackAudioClient *  gst_jack_audio_client_new            (const gchar *id, const gchar *server,
+                                                            jack_client_t *jclient,
+                                                            GstJackClientType type,
+                                                            void (*shutdown) (void *arg),
+                                                            JackProcessCallback    process,
+                                                            JackBufferSizeCallback buffer_size,
+                                                            JackSampleRateCallback sample_rate,
+							    gpointer user_data,
+							    jack_status_t *status);
+void                  gst_jack_audio_client_free           (GstJackAudioClient *client);
+
+jack_client_t *       gst_jack_audio_client_get_client     (GstJackAudioClient *client);
+
+gboolean              gst_jack_audio_client_set_active     (GstJackAudioClient *client, gboolean active);
+
+GstState              gst_jack_audio_client_get_transport_state (GstJackAudioClient *client);
+
+G_END_DECLS
+
+#endif /* __GST_JACK_AUDIO_CLIENT_H__ */
diff --git a/ext/jack/gstjackaudiosink.c b/ext/jack/gstjackaudiosink.c
new file mode 100755
index 0000000..3a83567
--- /dev/null
+++ b/ext/jack/gstjackaudiosink.c
@@ -0,0 +1,941 @@
+/* GStreamer
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ *
+ * gstjackaudiosink.c: jack audio sink implementation
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:element-jackaudiosink
+ * @see_also: #GstAudioBaseSink, #GstAudioRingBuffer
+ *
+ * A Sink that outputs data to Jack ports.
+ * 
+ * It will create N Jack ports named out_&lt;name&gt;_&lt;num&gt; where 
+ * &lt;name&gt; is the element name and &lt;num&gt; is starting from 1.
+ * Each port corresponds to a gstreamer channel.
+ * 
+ * The samplerate as exposed on the caps is always the same as the samplerate of
+ * the jack server.
+ * 
+ * When the #GstJackAudioSink:connect property is set to auto, this element
+ * will try to connect each output port to a random physical jack input pin. In
+ * this mode, the sink will expose the number of physical channels on its pad
+ * caps.
+ * 
+ * When the #GstJackAudioSink:connect property is set to none, the element will
+ * accept any number of input channels and will create (but not connect) an
+ * output port for each channel.
+ * 
+ * The element will generate an error when the Jack server is shut down when it
+ * was PAUSED or PLAYING. This element does not support dynamic rate and buffer
+ * size changes at runtime.
+ * 
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch-1.0 audiotestsrc ! jackaudiosink
+ * ]| Play a sine wave to using jack.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst-i18n-plugin.h>
+#include <stdlib.h>
+#include <string.h>
+#include <gst/audio/audio.h>
+
+#include "gstjackaudiosink.h"
+#include "gstjackringbuffer.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_sink_debug);
+#define GST_CAT_DEFAULT gst_jack_audio_sink_debug
+
+static gboolean
+gst_jack_audio_sink_allocate_channels (GstJackAudioSink * sink, gint channels)
+{
+  jack_client_t *client;
+
+  client = gst_jack_audio_client_get_client (sink->client);
+
+  /* remove ports we don't need */
+  while (sink->port_count > channels) {
+    jack_port_unregister (client, sink->ports[--sink->port_count]);
+  }
+
+  /* alloc enough output ports */
+  sink->ports = g_realloc (sink->ports, sizeof (jack_port_t *) * channels);
+  sink->buffers = g_realloc (sink->buffers, sizeof (sample_t *) * channels);
+
+  /* create an output port for each channel */
+  while (sink->port_count < channels) {
+    gchar *name;
+
+    /* port names start from 1 and are local to the element */
+    name =
+        g_strdup_printf ("out_%s_%d", GST_ELEMENT_NAME (sink),
+        sink->port_count + 1);
+    sink->ports[sink->port_count] =
+        jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
+        JackPortIsOutput, 0);
+    if (sink->ports[sink->port_count] == NULL)
+      return FALSE;
+
+    sink->port_count++;
+
+    g_free (name);
+  }
+  return TRUE;
+}
+
+static void
+gst_jack_audio_sink_free_channels (GstJackAudioSink * sink)
+{
+  gint res, i = 0;
+  jack_client_t *client;
+
+  client = gst_jack_audio_client_get_client (sink->client);
+
+  /* get rid of all ports */
+  while (sink->port_count) {
+    GST_LOG_OBJECT (sink, "unregister port %d", i);
+    if ((res = jack_port_unregister (client, sink->ports[i++]))) {
+      GST_DEBUG_OBJECT (sink, "unregister of port failed (%d)", res);
+    }
+    sink->port_count--;
+  }
+  g_free (sink->ports);
+  sink->ports = NULL;
+  g_free (sink->buffers);
+  sink->buffers = NULL;
+}
+
+/* ringbuffer abstract base class */
+static GType
+gst_jack_ring_buffer_get_type (void)
+{
+  static volatile gsize ringbuffer_type = 0;
+
+  if (g_once_init_enter (&ringbuffer_type)) {
+    static const GTypeInfo ringbuffer_info = {
+      sizeof (GstJackRingBufferClass),
+      NULL,
+      NULL,
+      (GClassInitFunc) gst_jack_ring_buffer_class_init,
+      NULL,
+      NULL,
+      sizeof (GstJackRingBuffer),
+      0,
+      (GInstanceInitFunc) gst_jack_ring_buffer_init,
+      NULL
+    };
+    GType tmp = g_type_register_static (GST_TYPE_AUDIO_RING_BUFFER,
+        "GstJackAudioSinkRingBuffer", &ringbuffer_info, 0);
+    g_once_init_leave (&ringbuffer_type, tmp);
+  }
+
+  return (GType) ringbuffer_type;
+}
+
+static void
+gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
+{
+  GstAudioRingBufferClass *gstringbuffer_class;
+
+  gstringbuffer_class = (GstAudioRingBufferClass *) klass;
+
+  ring_parent_class = g_type_class_peek_parent (klass);
+
+  gstringbuffer_class->open_device =
+      GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
+  gstringbuffer_class->close_device =
+      GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
+  gstringbuffer_class->acquire =
+      GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
+  gstringbuffer_class->release =
+      GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
+  gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
+  gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
+  gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
+  gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
+
+  gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
+}
+
+/* this is the callback of jack. This should RT-safe.
+ */
+static int
+jack_process_cb (jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioSink *sink;
+  GstAudioRingBuffer *buf;
+  gint readseg, len;
+  guint8 *readptr;
+  gint i, j, flen, channels;
+  sample_t *data;
+
+  buf = GST_AUDIO_RING_BUFFER_CAST (arg);
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
+
+  /* get target buffers */
+  for (i = 0; i < channels; i++) {
+    sink->buffers[i] =
+        (sample_t *) jack_port_get_buffer (sink->ports[i], nframes);
+  }
+
+  if (gst_audio_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) {
+    flen = len / channels;
+
+    /* the number of samples must be exactly the segment size */
+    if (nframes * sizeof (sample_t) != flen)
+      goto wrong_size;
+
+    GST_DEBUG_OBJECT (sink, "copy %d frames: %p, %d bytes, %d channels",
+        nframes, readptr, flen, channels);
+    data = (sample_t *) readptr;
+
+    /* the samples in the ringbuffer have the channels interleaved, we need to
+     * deinterleave into the jack target buffers */
+    for (i = 0; i < nframes; i++) {
+      for (j = 0; j < channels; j++) {
+        sink->buffers[j][i] = *data++;
+      }
+    }
+
+    /* clear written samples in the ringbuffer */
+    gst_audio_ring_buffer_clear (buf, readseg);
+
+    /* we wrote one segment */
+    gst_audio_ring_buffer_advance (buf, 1);
+  } else {
+    GST_DEBUG_OBJECT (sink, "write %d frames silence", nframes);
+    /* We are not allowed to read from the ringbuffer, write silence to all
+     * jack output buffers */
+    for (i = 0; i < channels; i++) {
+      memset (sink->buffers[i], 0, nframes * sizeof (sample_t));
+    }
+  }
+  return 0;
+
+  /* ERRORS */
+wrong_size:
+  {
+    GST_ERROR_OBJECT (sink, "nbytes (%d) != flen (%d)",
+        (gint) (nframes * sizeof (sample_t)), flen);
+    return 1;
+  }
+}
+
+/* we error out */
+static int
+jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioSink *sink;
+  GstJackRingBuffer *abuf;
+
+  abuf = GST_JACK_RING_BUFFER_CAST (arg);
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
+
+  if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
+    goto not_supported;
+
+  return 0;
+
+  /* ERRORS */
+not_supported:
+  {
+    GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
+        (NULL), ("Jack changed the sample rate, which is not supported"));
+    return 1;
+  }
+}
+
+/* we error out */
+static int
+jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioSink *sink;
+  GstJackRingBuffer *abuf;
+
+  abuf = GST_JACK_RING_BUFFER_CAST (arg);
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
+
+  if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
+    goto not_supported;
+
+  return 0;
+
+  /* ERRORS */
+not_supported:
+  {
+    GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS,
+        (NULL), ("Jack changed the buffer size, which is not supported"));
+    return 1;
+  }
+}
+
+static void
+jack_shutdown_cb (void *arg)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (arg));
+
+  GST_DEBUG_OBJECT (sink, "shutdown");
+
+  GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
+      (NULL), ("Jack server shutdown"));
+}
+
+static void
+gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
+    GstJackRingBufferClass * g_class)
+{
+  buf->channels = -1;
+  buf->buffer_size = -1;
+  buf->sample_rate = -1;
+}
+
+/* the _open_device method should make a connection with the server
+ */
+static gboolean
+gst_jack_ring_buffer_open_device (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+  jack_status_t status = 0;
+  const gchar *name;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "open");
+
+  if (sink->client_name) {
+    name = sink->client_name;
+  } else {
+    name = g_get_application_name ();
+  }
+  if (!name)
+    name = "GStreamer";
+
+  sink->client = gst_jack_audio_client_new (name, sink->server,
+      sink->jclient,
+      GST_JACK_CLIENT_SINK,
+      jack_shutdown_cb,
+      jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
+  if (sink->client == NULL)
+    goto could_not_open;
+
+  GST_DEBUG_OBJECT (sink, "opened");
+
+  return TRUE;
+
+  /* ERRORS */
+could_not_open:
+  {
+    if (status & JackServerFailed) {
+      GST_ELEMENT_ERROR (sink, RESOURCE, NOT_FOUND,
+          (_("Jack server not found")),
+          ("Cannot connect to the Jack server (status %d)", status));
+    } else {
+      GST_ELEMENT_ERROR (sink, RESOURCE, OPEN_WRITE,
+          (NULL), ("Jack client open error (status %d)", status));
+    }
+    return FALSE;
+  }
+}
+
+/* close the connection with the server
+ */
+static gboolean
+gst_jack_ring_buffer_close_device (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "close");
+
+  gst_jack_audio_sink_free_channels (sink);
+  gst_jack_audio_client_free (sink->client);
+  sink->client = NULL;
+
+  return TRUE;
+}
+
+/* allocate a buffer and setup resources to process the audio samples of
+ * the format as specified in @spec.
+ *
+ * We allocate N jack ports, one for each channel. If we are asked to
+ * automatically make a connection with physical ports, we connect as many
+ * ports as there are physical ports, leaving leftover ports unconnected.
+ *
+ * It is assumed that samplerate and number of channels are acceptable since our
+ * getcaps method will always provide correct values. If unacceptable caps are
+ * received for some reason, we fail here.
+ */
+static gboolean
+gst_jack_ring_buffer_acquire (GstAudioRingBuffer * buf,
+    GstAudioRingBufferSpec * spec)
+{
+  GstJackAudioSink *sink;
+  GstJackRingBuffer *abuf;
+  const char **ports;
+  gint sample_rate, buffer_size;
+  gint i, rate, bpf, channels, res;
+  jack_client_t *client;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+  abuf = GST_JACK_RING_BUFFER_CAST (buf);
+
+  GST_DEBUG_OBJECT (sink, "acquire");
+
+  client = gst_jack_audio_client_get_client (sink->client);
+
+  rate = GST_AUDIO_INFO_RATE (&spec->info);
+
+  /* sample rate must be that of the server */
+  sample_rate = jack_get_sample_rate (client);
+  if (sample_rate != rate)
+    goto wrong_samplerate;
+
+  channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
+  bpf = GST_AUDIO_INFO_BPF (&spec->info);
+
+  if (!gst_jack_audio_sink_allocate_channels (sink, channels))
+    goto out_of_ports;
+
+  buffer_size = jack_get_buffer_size (client);
+
+  /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
+   * for all channels  */
+  spec->segsize = buffer_size * sizeof (gfloat) * channels;
+  spec->latency_time = gst_util_uint64_scale (spec->segsize,
+      (GST_SECOND / GST_USECOND), rate * bpf);
+  /* segtotal based on buffer-time latency */
+  spec->segtotal = spec->buffer_time / spec->latency_time;
+  if (spec->segtotal < 2) {
+    spec->segtotal = 2;
+    spec->buffer_time = spec->latency_time * spec->segtotal;
+  }
+
+  GST_DEBUG_OBJECT (sink, "buffer time: %" G_GINT64_FORMAT " usec",
+      spec->buffer_time);
+  GST_DEBUG_OBJECT (sink, "latency time: %" G_GINT64_FORMAT " usec",
+      spec->latency_time);
+  GST_DEBUG_OBJECT (sink, "buffer_size %d, segsize %d, segtotal %d",
+      buffer_size, spec->segsize, spec->segtotal);
+
+  /* allocate the ringbuffer memory now */
+  buf->size = spec->segtotal * spec->segsize;
+  buf->memory = g_malloc0 (buf->size);
+
+  if ((res = gst_jack_audio_client_set_active (sink->client, TRUE)))
+    goto could_not_activate;
+
+  /* if we need to automatically connect the ports, do so now. We must do this
+   * after activating the client. */
+  if (sink->connect == GST_JACK_CONNECT_AUTO
+      || sink->connect == GST_JACK_CONNECT_AUTO_FORCED) {
+    /* find all the physical input ports. A physical input port is a port
+     * associated with a hardware device. Someone needs connect to a physical
+     * port in order to hear something. */
+    ports = jack_get_ports (client, NULL, NULL,
+        JackPortIsPhysical | JackPortIsInput);
+    if (ports == NULL) {
+      /* no ports? fine then we don't do anything except for posting a warning
+       * message. */
+      GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
+          ("No physical input ports found, leaving ports unconnected"));
+      goto done;
+    }
+
+    for (i = 0; i < channels; i++) {
+      /* stop when all input ports are exhausted */
+      if (ports[i] == NULL) {
+        /* post a warning that we could not connect all ports */
+        GST_ELEMENT_WARNING (sink, RESOURCE, NOT_FOUND, (NULL),
+            ("No more physical ports, leaving some ports unconnected"));
+        break;
+      }
+      GST_DEBUG_OBJECT (sink, "try connecting to %s",
+          jack_port_name (sink->ports[i]));
+      /* connect the port to a physical port */
+      res = jack_connect (client, jack_port_name (sink->ports[i]), ports[i]);
+      if (res != 0 && res != EEXIST)
+        goto cannot_connect;
+    }
+    free (ports);
+  }
+done:
+
+  abuf->sample_rate = sample_rate;
+  abuf->buffer_size = buffer_size;
+  abuf->channels = channels;
+
+  return TRUE;
+
+  /* ERRORS */
+wrong_samplerate:
+  {
+    GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+        ("Wrong samplerate, server is running at %d and we received %d",
+            sample_rate, rate));
+    return FALSE;
+  }
+out_of_ports:
+  {
+    GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+        ("Cannot allocate more Jack ports"));
+    return FALSE;
+  }
+could_not_activate:
+  {
+    GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+        ("Could not activate client (%d:%s)", res, g_strerror (res)));
+    return FALSE;
+  }
+cannot_connect:
+  {
+    GST_ELEMENT_ERROR (sink, RESOURCE, SETTINGS, (NULL),
+        ("Could not connect output ports to physical ports (%d:%s)",
+            res, g_strerror (res)));
+    free (ports);
+    return FALSE;
+  }
+}
+
+/* function is called with LOCK */
+static gboolean
+gst_jack_ring_buffer_release (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+  GstJackRingBuffer *abuf;
+  gint res;
+
+  abuf = GST_JACK_RING_BUFFER_CAST (buf);
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "release");
+
+  if ((res = gst_jack_audio_client_set_active (sink->client, FALSE))) {
+    /* we only warn, this means the server is probably shut down and the client
+     * is gone anyway. */
+    GST_ELEMENT_WARNING (sink, RESOURCE, CLOSE, (NULL),
+        ("Could not deactivate Jack client (%d)", res));
+  }
+
+  abuf->channels = -1;
+  abuf->buffer_size = -1;
+  abuf->sample_rate = -1;
+
+  /* free the buffer */
+  g_free (buf->memory);
+  buf->memory = NULL;
+
+  return TRUE;
+}
+
+static gboolean
+gst_jack_ring_buffer_start (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "start");
+
+  if (sink->transport & GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (sink->client);
+    jack_transport_start (client);
+  }
+
+  return TRUE;
+}
+
+static gboolean
+gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "pause");
+
+  if (sink->transport & GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (sink->client);
+    jack_transport_stop (client);
+  }
+
+  return TRUE;
+}
+
+static gboolean
+gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (sink, "stop");
+
+  if (sink->transport & GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (sink->client);
+    jack_transport_stop (client);
+  }
+
+  return TRUE;
+}
+
+#if defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)
+static guint
+gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+  guint i, res = 0;
+  jack_latency_range_t range;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+
+  for (i = 0; i < sink->port_count; i++) {
+    jack_port_get_latency_range (sink->ports[i], JackPlaybackLatency, &range);
+    if (range.max > res)
+      res = range.max;
+  }
+
+  GST_LOG_OBJECT (sink, "delay %u", res);
+
+  return res;
+}
+#else /* !(defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)) */
+static guint
+gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSink *sink;
+  guint i, res = 0;
+  guint latency;
+  jack_client_t *client;
+
+  sink = GST_JACK_AUDIO_SINK (GST_OBJECT_PARENT (buf));
+  client = gst_jack_audio_client_get_client (sink->client);
+
+  for (i = 0; i < sink->port_count; i++) {
+    latency = jack_port_get_total_latency (client, sink->ports[i]);
+    if (latency > res)
+      res = latency;
+  }
+
+  GST_LOG_OBJECT (sink, "delay %u", res);
+
+  return res;
+}
+#endif
+
+static GstStaticPadTemplate jackaudiosink_sink_factory =
+GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " GST_JACK_FORMAT_STR ", "
+        "layout = (string) interleaved, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
+    );
+
+/* AudioSink signals and args */
+enum
+{
+  /* FILL ME */
+  SIGNAL_LAST
+};
+
+#define DEFAULT_PROP_CONNECT 		GST_JACK_CONNECT_AUTO
+#define DEFAULT_PROP_SERVER 		NULL
+#define DEFAULT_PROP_CLIENT_NAME	NULL
+#define DEFAULT_PROP_TRANSPORT	GST_JACK_TRANSPORT_AUTONOMOUS
+
+enum
+{
+  PROP_0,
+  PROP_CONNECT,
+  PROP_SERVER,
+  PROP_CLIENT,
+  PROP_CLIENT_NAME,
+  PROP_TRANSPORT,
+  PROP_LAST
+};
+
+#define gst_jack_audio_sink_parent_class parent_class
+G_DEFINE_TYPE (GstJackAudioSink, gst_jack_audio_sink, GST_TYPE_AUDIO_BASE_SINK);
+
+static void gst_jack_audio_sink_dispose (GObject * object);
+static void gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static GstCaps *gst_jack_audio_sink_getcaps (GstBaseSink * bsink,
+    GstCaps * filter);
+static GstAudioRingBuffer
+    * gst_jack_audio_sink_create_ringbuffer (GstAudioBaseSink * sink);
+
+static void
+gst_jack_audio_sink_class_init (GstJackAudioSinkClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseSinkClass *gstbasesink_class;
+  GstAudioBaseSinkClass *gstaudiobasesink_class;
+
+  GST_DEBUG_CATEGORY_INIT (gst_jack_audio_sink_debug, "jacksink", 0,
+      "jacksink element");
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasesink_class = (GstBaseSinkClass *) klass;
+  gstaudiobasesink_class = (GstAudioBaseSinkClass *) klass;
+
+  gobject_class->dispose = gst_jack_audio_sink_dispose;
+  gobject_class->get_property = gst_jack_audio_sink_get_property;
+  gobject_class->set_property = gst_jack_audio_sink_set_property;
+
+  g_object_class_install_property (gobject_class, PROP_CONNECT,
+      g_param_spec_enum ("connect", "Connect",
+          "Specify how the output ports will be connected",
+          GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_SERVER,
+      g_param_spec_string ("server", "Server",
+          "The Jack server to connect to (NULL = default)",
+          DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstJackAudioSink:client-name:
+   *
+   * The client name to use.
+   */
+  g_object_class_install_property (gobject_class, PROP_CLIENT_NAME,
+      g_param_spec_string ("client-name", "Client name",
+          "The client name of the Jack instance (NULL = default)",
+          DEFAULT_PROP_CLIENT_NAME,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_CLIENT,
+      g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
+          GST_TYPE_JACK_CLIENT,
+          GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
+          G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstJackAudioSink:transport:
+   *
+   * The jack transport behaviour for the client.
+   */
+  g_object_class_install_property (gobject_class, PROP_TRANSPORT,
+      g_param_spec_flags ("transport", "Transport mode",
+          "Jack transport behaviour of the client",
+          GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_element_class_set_static_metadata (gstelement_class, "Audio Sink (Jack)",
+      "Sink/Audio", "Output audio to a JACK server",
+      "Wim Taymans <wim.taymans@gmail.com>");
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&jackaudiosink_sink_factory));
+
+  gstbasesink_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_sink_getcaps);
+
+  gstaudiobasesink_class->create_ringbuffer =
+      GST_DEBUG_FUNCPTR (gst_jack_audio_sink_create_ringbuffer);
+
+  /* ref class from a thread-safe context to work around missing bit of
+   * thread-safety in GObject */
+  g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
+
+  gst_jack_audio_client_init ();
+}
+
+static void
+gst_jack_audio_sink_init (GstJackAudioSink * sink)
+{
+  sink->connect = DEFAULT_PROP_CONNECT;
+  sink->server = g_strdup (DEFAULT_PROP_SERVER);
+  sink->jclient = NULL;
+  sink->ports = NULL;
+  sink->port_count = 0;
+  sink->buffers = NULL;
+  sink->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
+  sink->transport = DEFAULT_PROP_TRANSPORT;
+}
+
+static void
+gst_jack_audio_sink_dispose (GObject * object)
+{
+  GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (object);
+
+  gst_caps_replace (&sink->caps, NULL);
+
+  if (sink->client_name != NULL) {
+    g_free (sink->client_name);
+    sink->client_name = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_jack_audio_sink_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (object);
+
+  switch (prop_id) {
+    case PROP_CLIENT_NAME:
+      g_free (sink->client_name);
+      sink->client_name = g_value_dup_string (value);
+      break;
+    case PROP_CONNECT:
+      sink->connect = g_value_get_enum (value);
+      break;
+    case PROP_SERVER:
+      g_free (sink->server);
+      sink->server = g_value_dup_string (value);
+      break;
+    case PROP_CLIENT:
+      if (GST_STATE (sink) == GST_STATE_NULL ||
+          GST_STATE (sink) == GST_STATE_READY) {
+        sink->jclient = g_value_get_boxed (value);
+      }
+      break;
+    case PROP_TRANSPORT:
+      sink->transport = g_value_get_flags (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_jack_audio_sink_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstJackAudioSink *sink;
+
+  sink = GST_JACK_AUDIO_SINK (object);
+
+  switch (prop_id) {
+    case PROP_CLIENT_NAME:
+      g_value_set_string (value, sink->client_name);
+      break;
+    case PROP_CONNECT:
+      g_value_set_enum (value, sink->connect);
+      break;
+    case PROP_SERVER:
+      g_value_set_string (value, sink->server);
+      break;
+    case PROP_CLIENT:
+      g_value_set_boxed (value, sink->jclient);
+      break;
+    case PROP_TRANSPORT:
+      g_value_set_flags (value, sink->transport);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstCaps *
+gst_jack_audio_sink_getcaps (GstBaseSink * bsink, GstCaps * filter)
+{
+  GstJackAudioSink *sink = GST_JACK_AUDIO_SINK (bsink);
+  const char **ports;
+  gint min, max;
+  gint rate;
+  jack_client_t *client;
+
+  if (sink->client == NULL)
+    goto no_client;
+
+  client = gst_jack_audio_client_get_client (sink->client);
+
+  if (sink->connect == GST_JACK_CONNECT_AUTO) {
+    /* get a port count, this is the number of channels we can automatically
+     * connect. */
+    ports = jack_get_ports (client, NULL, NULL,
+        JackPortIsPhysical | JackPortIsInput);
+    max = 0;
+    if (ports != NULL) {
+      for (; ports[max]; max++);
+      free (ports);
+    } else
+      max = 0;
+  } else {
+    /* we allow any number of pads, something else is going to connect the
+     * pads. */
+    max = G_MAXINT;
+  }
+  min = MIN (1, max);
+
+  rate = jack_get_sample_rate (client);
+
+  GST_DEBUG_OBJECT (sink, "got %d-%d ports, samplerate: %d", min, max, rate);
+
+  if (!sink->caps) {
+    sink->caps = gst_caps_new_simple ("audio/x-raw",
+        "format", G_TYPE_STRING, GST_JACK_FORMAT_STR,
+        "layout", G_TYPE_STRING, "interleaved",
+        "rate", G_TYPE_INT, rate,
+        "channels", GST_TYPE_INT_RANGE, min, max, NULL);
+  }
+  GST_INFO_OBJECT (sink, "returning caps %" GST_PTR_FORMAT, sink->caps);
+
+  return gst_caps_ref (sink->caps);
+
+  /* ERRORS */
+no_client:
+  {
+    GST_DEBUG_OBJECT (sink, "device not open, using template caps");
+    /* base class will get template caps for us when we return NULL */
+    return NULL;
+  }
+}
+
+static GstAudioRingBuffer *
+gst_jack_audio_sink_create_ringbuffer (GstAudioBaseSink * sink)
+{
+  GstAudioRingBuffer *buffer;
+
+  buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
+  GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer);
+
+  return buffer;
+}
diff --git a/ext/jack/gstjackaudiosink.h b/ext/jack/gstjackaudiosink.h
new file mode 100644
index 0000000..b2377c4
--- /dev/null
+++ b/ext/jack/gstjackaudiosink.h
@@ -0,0 +1,81 @@
+/* GStreamer
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ *
+ * gstjacksink.h:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_JACK_AUDIO_SINK_H__
+#define __GST_JACK_AUDIO_SINK_H__
+
+#include <jack/jack.h>
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiobasesink.h>
+
+#include "gstjack.h"
+#include "gstjackaudioclient.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_JACK_AUDIO_SINK             (gst_jack_audio_sink_get_type())
+#define GST_JACK_AUDIO_SINK(obj)             (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSink))
+#define GST_JACK_AUDIO_SINK_CLASS(klass)     (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass))
+#define GST_JACK_AUDIO_SINK_GET_CLASS(obj)   (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SINK,GstJackAudioSinkClass))
+#define GST_IS_JACK_AUDIO_SINK(obj)          (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SINK))
+#define GST_IS_JACK_AUDIO_SINK_CLASS(klass)  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SINK))
+
+typedef struct _GstJackAudioSink GstJackAudioSink;
+typedef struct _GstJackAudioSinkClass GstJackAudioSinkClass;
+
+/**
+ * GstJackAudioSink:
+ * 
+ * Opaque #GstJackAudioSink.
+ */
+struct _GstJackAudioSink {
+  GstAudioBaseSink element;
+
+  /*< private >*/
+  /* cached caps */
+  GstCaps         *caps;
+
+  /* properties */
+  GstJackConnect   connect;
+  gchar           *server;
+  jack_client_t   *jclient;
+  gchar           *client_name;
+  guint            transport;
+
+  /* our client */
+  GstJackAudioClient *client;
+
+  /* our ports */
+  jack_port_t    **ports;
+  int              port_count;
+  sample_t       **buffers;
+};
+
+struct _GstJackAudioSinkClass {
+  GstAudioBaseSinkClass parent_class;
+};
+
+GType gst_jack_audio_sink_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_JACK_AUDIO_SINK_H__ */
diff --git a/ext/jack/gstjackaudiosrc.c b/ext/jack/gstjackaudiosrc.c
new file mode 100755
index 0000000..5a3bfb5
--- /dev/null
+++ b/ext/jack/gstjackaudiosrc.c
@@ -0,0 +1,961 @@
+/* GStreamer
+ * Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
+ * 
+ * Permission is hereby granted, free of charge, to any person obtaining a
+ * copy of this software and associated documentation files (the "Software"),
+ * to deal in the Software without restriction, including without limitation
+ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
+ * and/or sell copies of the Software, and to permit persons to whom the
+ * Software is furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
+ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
+ * DEALINGS IN THE SOFTWARE.
+ *
+ * Alternatively, the contents of this file may be used under the
+ * GNU Lesser General Public License Version 2.1 (the "LGPL"), in
+ * which case the following provisions apply instead of the ones
+ * mentioned above:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+/**
+ * SECTION:element-jackaudiosrc
+ * @see_also: #GstAudioBaseSrc, #GstAudioRingBuffer
+ *
+ * A Src that inputs data from Jack ports.
+ * 
+ * It will create N Jack ports named in_&lt;name&gt;_&lt;num&gt; where 
+ * &lt;name&gt; is the element name and &lt;num&gt; is starting from 1.
+ * Each port corresponds to a gstreamer channel.
+ * 
+ * The samplerate as exposed on the caps is always the same as the samplerate of
+ * the jack server.
+ * 
+ * When the #GstJackAudioSrc:connect property is set to auto, this element
+ * will try to connect each input port to a random physical jack output pin. 
+ * 
+ * When the #GstJackAudioSrc:connect property is set to none, the element will
+ * accept any number of output channels and will create (but not connect) an
+ * input port for each channel.
+ * 
+ * The element will generate an error when the Jack server is shut down when it
+ * was PAUSED or PLAYING. This element does not support dynamic rate and buffer
+ * size changes at runtime.
+ * 
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch-1.0 jackaudiosrc connect=0 ! jackaudiosink connect=0
+ * ]| Get audio input into gstreamer from jack.
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include "config.h"
+#endif
+
+#include <gst/gst-i18n-plugin.h>
+#include <stdlib.h>
+#include <string.h>
+
+#include <gst/audio/audio.h>
+
+#include "gstjackaudiosrc.h"
+#include "gstjackringbuffer.h"
+#include "gstjackutil.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_jack_audio_src_debug);
+#define GST_CAT_DEFAULT gst_jack_audio_src_debug
+
+static gboolean
+gst_jack_audio_src_allocate_channels (GstJackAudioSrc * src, gint channels)
+{
+  jack_client_t *client;
+
+  client = gst_jack_audio_client_get_client (src->client);
+
+  /* remove ports we don't need */
+  while (src->port_count > channels)
+    jack_port_unregister (client, src->ports[--src->port_count]);
+
+  /* alloc enough input ports */
+  src->ports = g_realloc (src->ports, sizeof (jack_port_t *) * channels);
+  src->buffers = g_realloc (src->buffers, sizeof (sample_t *) * channels);
+
+  /* create an input port for each channel */
+  while (src->port_count < channels) {
+    gchar *name;
+
+    /* port names start from 1 and are local to the element */
+    name =
+        g_strdup_printf ("in_%s_%d", GST_ELEMENT_NAME (src),
+        src->port_count + 1);
+    src->ports[src->port_count] =
+        jack_port_register (client, name, JACK_DEFAULT_AUDIO_TYPE,
+        JackPortIsInput, 0);
+    if (src->ports[src->port_count] == NULL)
+      return FALSE;
+
+    src->port_count++;
+
+    g_free (name);
+  }
+  return TRUE;
+}
+
+static void
+gst_jack_audio_src_free_channels (GstJackAudioSrc * src)
+{
+  gint res, i = 0;
+  jack_client_t *client;
+
+  client = gst_jack_audio_client_get_client (src->client);
+
+  /* get rid of all ports */
+  while (src->port_count) {
+    GST_LOG_OBJECT (src, "unregister port %d", i);
+    if ((res = jack_port_unregister (client, src->ports[i++])))
+      GST_DEBUG_OBJECT (src, "unregister of port failed (%d)", res);
+
+    src->port_count--;
+  }
+  g_free (src->ports);
+  src->ports = NULL;
+  g_free (src->buffers);
+  src->buffers = NULL;
+}
+
+/* ringbuffer abstract base class */
+static GType
+gst_jack_ring_buffer_get_type (void)
+{
+  static volatile gsize ringbuffer_type = 0;
+
+  if (g_once_init_enter (&ringbuffer_type)) {
+    static const GTypeInfo ringbuffer_info = { sizeof (GstJackRingBufferClass),
+      NULL,
+      NULL,
+      (GClassInitFunc) gst_jack_ring_buffer_class_init,
+      NULL,
+      NULL,
+      sizeof (GstJackRingBuffer),
+      0,
+      (GInstanceInitFunc) gst_jack_ring_buffer_init,
+      NULL
+    };
+    GType tmp = g_type_register_static (GST_TYPE_AUDIO_RING_BUFFER,
+        "GstJackAudioSrcRingBuffer", &ringbuffer_info, 0);
+    g_once_init_leave (&ringbuffer_type, tmp);
+  }
+
+  return (GType) ringbuffer_type;
+}
+
+static void
+gst_jack_ring_buffer_class_init (GstJackRingBufferClass * klass)
+{
+  GstAudioRingBufferClass *gstringbuffer_class;
+
+  gstringbuffer_class = (GstAudioRingBufferClass *) klass;
+
+  ring_parent_class = g_type_class_peek_parent (klass);
+
+  gstringbuffer_class->open_device =
+      GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_open_device);
+  gstringbuffer_class->close_device =
+      GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_close_device);
+  gstringbuffer_class->acquire =
+      GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_acquire);
+  gstringbuffer_class->release =
+      GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_release);
+  gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
+  gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_pause);
+  gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_start);
+  gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_stop);
+
+  gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_jack_ring_buffer_delay);
+}
+
+/* this is the callback of jack. This should be RT-safe.
+ * Writes samples from the jack input port's buffer to the gst ring buffer.
+ */
+static int
+jack_process_cb (jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioSrc *src;
+  GstAudioRingBuffer *buf;
+  gint len;
+  guint8 *writeptr;
+  gint writeseg;
+  gint channels, i, j, flen;
+  sample_t *data;
+
+  buf = GST_AUDIO_RING_BUFFER_CAST (arg);
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+  channels = GST_AUDIO_INFO_CHANNELS (&buf->spec.info);
+
+  /* get input buffers */
+  for (i = 0; i < channels; i++)
+    src->buffers[i] =
+        (sample_t *) jack_port_get_buffer (src->ports[i], nframes);
+
+  if (gst_audio_ring_buffer_prepare_read (buf, &writeseg, &writeptr, &len)) {
+    flen = len / channels;
+
+    /* the number of samples must be exactly the segment size */
+    if (nframes * sizeof (sample_t) != flen)
+      goto wrong_size;
+
+    /* the samples in the jack input buffers have to be interleaved into the
+     * ringbuffer */
+    data = (sample_t *) writeptr;
+    for (i = 0; i < nframes; ++i)
+      for (j = 0; j < channels; ++j)
+        *data++ = src->buffers[j][i];
+
+    GST_DEBUG ("copy %d frames: %p, %d bytes, %d channels", nframes, writeptr,
+        len / channels, channels);
+
+    /* we wrote one segment */
+    gst_audio_ring_buffer_advance (buf, 1);
+  }
+  return 0;
+
+  /* ERRORS */
+wrong_size:
+  {
+    GST_ERROR_OBJECT (src, "nbytes (%d) != flen (%d)",
+        (gint) (nframes * sizeof (sample_t)), flen);
+    return 1;
+  }
+}
+
+/* we error out */
+static int
+jack_sample_rate_cb (jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioSrc *src;
+  GstJackRingBuffer *abuf;
+
+  abuf = GST_JACK_RING_BUFFER_CAST (arg);
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
+
+  if (abuf->sample_rate != -1 && abuf->sample_rate != nframes)
+    goto not_supported;
+
+  return 0;
+
+  /* ERRORS */
+not_supported:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
+        (NULL), ("Jack changed the sample rate, which is not supported"));
+    return 1;
+  }
+}
+
+/* we error out */
+static int
+jack_buffer_size_cb (jack_nframes_t nframes, void *arg)
+{
+  GstJackAudioSrc *src;
+  GstJackRingBuffer *abuf;
+
+  abuf = GST_JACK_RING_BUFFER_CAST (arg);
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
+
+  if (abuf->buffer_size != -1 && abuf->buffer_size != nframes)
+    goto not_supported;
+
+  return 0;
+
+  /* ERRORS */
+not_supported:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS,
+        (NULL), ("Jack changed the buffer size, which is not supported"));
+    return 1;
+  }
+}
+
+static void
+jack_shutdown_cb (void *arg)
+{
+  GstJackAudioSrc *src;
+
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (arg));
+
+  GST_DEBUG_OBJECT (src, "shutdown");
+
+  GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
+      (NULL), ("Jack server shutdown"));
+}
+
+static void
+gst_jack_ring_buffer_init (GstJackRingBuffer * buf,
+    GstJackRingBufferClass * g_class)
+{
+  buf->channels = -1;
+  buf->buffer_size = -1;
+  buf->sample_rate = -1;
+}
+
+/* the _open_device method should make a connection with the server
+*/
+static gboolean
+gst_jack_ring_buffer_open_device (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+  jack_status_t status = 0;
+  const gchar *name;
+
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (src, "open");
+
+  if (src->client_name) {
+    name = src->client_name;
+  } else {
+    name = g_get_application_name ();
+  }
+  if (!name)
+    name = "GStreamer";
+
+  src->client = gst_jack_audio_client_new (name, src->server,
+      src->jclient,
+      GST_JACK_CLIENT_SOURCE,
+      jack_shutdown_cb,
+      jack_process_cb, jack_buffer_size_cb, jack_sample_rate_cb, buf, &status);
+  if (src->client == NULL)
+    goto could_not_open;
+
+  GST_DEBUG_OBJECT (src, "opened");
+
+  return TRUE;
+
+  /* ERRORS */
+could_not_open:
+  {
+    if (status & (JackServerFailed | JackFailure)) {
+      GST_ELEMENT_ERROR (src, RESOURCE, NOT_FOUND,
+          (_("Jack server not found")),
+          ("Cannot connect to the Jack server (status %d)", status));
+    } else {
+      GST_ELEMENT_ERROR (src, RESOURCE, OPEN_READ,
+          (NULL), ("Jack client open error (status %d)", status));
+    }
+    return FALSE;
+  }
+}
+
+/* close the connection with the server
+*/
+static gboolean
+gst_jack_ring_buffer_close_device (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (src, "close");
+
+  gst_jack_audio_src_free_channels (src);
+  gst_jack_audio_client_free (src->client);
+  src->client = NULL;
+
+  return TRUE;
+}
+
+
+/* allocate a buffer and setup resources to process the audio samples of
+ * the format as specified in @spec.
+ *
+ * We allocate N jack ports, one for each channel. If we are asked to
+ * automatically make a connection with physical ports, we connect as many
+ * ports as there are physical ports, leaving leftover ports unconnected.
+ *
+ * It is assumed that samplerate and number of channels are acceptable since our
+ * getcaps method will always provide correct values. If unacceptable caps are
+ * received for some reason, we fail here.
+ */
+static gboolean
+gst_jack_ring_buffer_acquire (GstAudioRingBuffer * buf,
+    GstAudioRingBufferSpec * spec)
+{
+  GstJackAudioSrc *src;
+  GstJackRingBuffer *abuf;
+  const char **ports;
+  gint sample_rate, buffer_size;
+  gint i, bpf, rate, channels, res;
+  jack_client_t *client;
+
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+  abuf = GST_JACK_RING_BUFFER_CAST (buf);
+
+  GST_DEBUG_OBJECT (src, "acquire");
+
+  client = gst_jack_audio_client_get_client (src->client);
+
+  rate = GST_AUDIO_INFO_RATE (&spec->info);
+
+  /* sample rate must be that of the server */
+  sample_rate = jack_get_sample_rate (client);
+  if (sample_rate != rate)
+    goto wrong_samplerate;
+
+  bpf = GST_AUDIO_INFO_BPF (&spec->info);
+  channels = GST_AUDIO_INFO_CHANNELS (&spec->info);
+
+  if (!gst_jack_audio_src_allocate_channels (src, channels))
+    goto out_of_ports;
+
+  gst_jack_set_layout (buf, spec);
+
+  buffer_size = jack_get_buffer_size (client);
+
+  /* the segment size in bytes, this is large enough to hold a buffer of 32bit floats
+   * for all channels  */
+  spec->segsize = buffer_size * sizeof (gfloat) * channels;
+  spec->latency_time = gst_util_uint64_scale (spec->segsize,
+      (GST_SECOND / GST_USECOND), rate * bpf);
+  /* segtotal based on buffer-time latency */
+  spec->segtotal = spec->buffer_time / spec->latency_time;
+  if (spec->segtotal < 2) {
+    spec->segtotal = 2;
+    spec->buffer_time = spec->latency_time * spec->segtotal;
+  }
+
+  GST_DEBUG_OBJECT (src, "buffer time: %" G_GINT64_FORMAT " usec",
+      spec->buffer_time);
+  GST_DEBUG_OBJECT (src, "latency time: %" G_GINT64_FORMAT " usec",
+      spec->latency_time);
+  GST_DEBUG_OBJECT (src, "buffer_size %d, segsize %d, segtotal %d",
+      buffer_size, spec->segsize, spec->segtotal);
+
+  /* allocate the ringbuffer memory now */
+  buf->size = spec->segtotal * spec->segsize;
+  buf->memory = g_malloc0 (buf->size);
+
+  if ((res = gst_jack_audio_client_set_active (src->client, TRUE)))
+    goto could_not_activate;
+
+  /* if we need to automatically connect the ports, do so now. We must do this
+   * after activating the client. */
+  if (src->connect == GST_JACK_CONNECT_AUTO
+      || src->connect == GST_JACK_CONNECT_AUTO_FORCED) {
+    /* find all the physical output ports. A physical output port is a port
+     * associated with a hardware device. Someone needs connect to a physical
+     * port in order to capture something. */
+    ports =
+        jack_get_ports (client, NULL, NULL,
+        JackPortIsPhysical | JackPortIsOutput);
+    if (ports == NULL) {
+      /* no ports? fine then we don't do anything except for posting a warning
+       * message. */
+      GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
+          ("No physical output ports found, leaving ports unconnected"));
+      goto done;
+    }
+
+    for (i = 0; i < channels; i++) {
+      /* stop when all output ports are exhausted */
+      if (ports[i] == NULL) {
+        /* post a warning that we could not connect all ports */
+        GST_ELEMENT_WARNING (src, RESOURCE, NOT_FOUND, (NULL),
+            ("No more physical ports, leaving some ports unconnected"));
+        break;
+      }
+      GST_DEBUG_OBJECT (src, "try connecting to %s",
+          jack_port_name (src->ports[i]));
+
+      /* connect the physical port to a port */
+      res = jack_connect (client, ports[i], jack_port_name (src->ports[i]));
+      if (res != 0 && res != EEXIST)
+        goto cannot_connect;
+    }
+    free (ports);
+  }
+done:
+
+  abuf->sample_rate = sample_rate;
+  abuf->buffer_size = buffer_size;
+  abuf->channels = channels;
+
+  return TRUE;
+
+  /* ERRORS */
+wrong_samplerate:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+        ("Wrong samplerate, server is running at %d and we received %d",
+            sample_rate, rate));
+    return FALSE;
+  }
+out_of_ports:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+        ("Cannot allocate more Jack ports"));
+    return FALSE;
+  }
+could_not_activate:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+        ("Could not activate client (%d:%s)", res, g_strerror (res)));
+    return FALSE;
+  }
+cannot_connect:
+  {
+    GST_ELEMENT_ERROR (src, RESOURCE, SETTINGS, (NULL),
+        ("Could not connect input ports to physical ports (%d:%s)",
+            res, g_strerror (res)));
+    free (ports);
+    return FALSE;
+  }
+}
+
+/* function is called with LOCK */
+static gboolean
+gst_jack_ring_buffer_release (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+  GstJackRingBuffer *abuf;
+  gint res;
+
+  abuf = GST_JACK_RING_BUFFER_CAST (buf);
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (src, "release");
+
+  if ((res = gst_jack_audio_client_set_active (src->client, FALSE))) {
+    /* we only warn, this means the server is probably shut down and the client
+     * is gone anyway. */
+    GST_ELEMENT_WARNING (src, RESOURCE, CLOSE, (NULL),
+        ("Could not deactivate Jack client (%d)", res));
+  }
+
+  abuf->channels = -1;
+  abuf->buffer_size = -1;
+  abuf->sample_rate = -1;
+
+  /* free the buffer */
+  g_free (buf->memory);
+  buf->memory = NULL;
+
+  return TRUE;
+}
+
+static gboolean
+gst_jack_ring_buffer_start (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (src, "start");
+
+  if (src->transport & GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (src->client);
+    jack_transport_start (client);
+  }
+
+  return TRUE;
+}
+
+static gboolean
+gst_jack_ring_buffer_pause (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (src, "pause");
+
+  if (src->transport & GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (src->client);
+    jack_transport_stop (client);
+  }
+
+  return TRUE;
+}
+
+static gboolean
+gst_jack_ring_buffer_stop (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+  GST_DEBUG_OBJECT (src, "stop");
+
+  if (src->transport & GST_JACK_TRANSPORT_MASTER) {
+    jack_client_t *client;
+
+    client = gst_jack_audio_client_get_client (src->client);
+    jack_transport_stop (client);
+  }
+
+  return TRUE;
+}
+
+#if defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)
+static guint
+gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+  guint i, res = 0;
+  jack_latency_range_t range;
+
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+  for (i = 0; i < src->port_count; i++) {
+    jack_port_get_latency_range (src->ports[i], JackCaptureLatency, &range);
+    if (range.max > res)
+      res = range.max;
+  }
+
+  GST_DEBUG_OBJECT (src, "delay %u", res);
+
+  return res;
+}
+#else /* !(defined (HAVE_JACK_0_120_1) || defined(HAVE_JACK_1_9_7)) */
+static guint
+gst_jack_ring_buffer_delay (GstAudioRingBuffer * buf)
+{
+  GstJackAudioSrc *src;
+  guint i, res = 0;
+  guint latency;
+  jack_client_t *client;
+
+  src = GST_JACK_AUDIO_SRC (GST_OBJECT_PARENT (buf));
+
+  client = gst_jack_audio_client_get_client (src->client);
+
+  for (i = 0; i < src->port_count; i++) {
+    latency = jack_port_get_total_latency (client, src->ports[i]);
+    if (latency > res)
+      res = latency;
+  }
+
+  GST_DEBUG_OBJECT (src, "delay %u", res);
+
+  return res;
+}
+#endif
+
+/* Audiosrc signals and args */
+enum
+{
+  /* FILL ME */
+  LAST_SIGNAL
+};
+
+#define DEFAULT_PROP_CONNECT 		GST_JACK_CONNECT_AUTO
+#define DEFAULT_PROP_SERVER 		NULL
+#define DEFAULT_PROP_CLIENT_NAME	NULL
+#define DEFAULT_PROP_TRANSPORT	GST_JACK_TRANSPORT_AUTONOMOUS
+
+enum
+{
+  PROP_0,
+  PROP_CONNECT,
+  PROP_SERVER,
+  PROP_CLIENT,
+  PROP_CLIENT_NAME,
+  PROP_TRANSPORT,
+  PROP_LAST
+};
+
+
+/* the capabilities of the inputs and outputs.
+ *
+ * describe the real formats here.
+ */
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " GST_JACK_FORMAT_STR ", "
+        "layout = (string) interleaved, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]")
+    );
+
+#define gst_jack_audio_src_parent_class parent_class
+G_DEFINE_TYPE (GstJackAudioSrc, gst_jack_audio_src, GST_TYPE_AUDIO_BASE_SRC);
+
+static void gst_jack_audio_src_dispose (GObject * object);
+static void gst_jack_audio_src_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_jack_audio_src_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static GstCaps *gst_jack_audio_src_getcaps (GstBaseSrc * bsrc,
+    GstCaps * filter);
+static GstAudioRingBuffer *gst_jack_audio_src_create_ringbuffer (GstAudioBaseSrc
+    * src);
+
+/* GObject vmethod implementations */
+
+/* initialize the jack_audio_src's class */
+static void
+gst_jack_audio_src_class_init (GstJackAudioSrcClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *gstelement_class;
+  GstBaseSrcClass *gstbasesrc_class;
+  GstAudioBaseSrcClass *gstaudiobasesrc_class;
+
+  GST_DEBUG_CATEGORY_INIT (gst_jack_audio_src_debug, "jacksrc", 0,
+      "jacksrc element");
+
+  gobject_class = (GObjectClass *) klass;
+  gstelement_class = (GstElementClass *) klass;
+  gstbasesrc_class = (GstBaseSrcClass *) klass;
+  gstaudiobasesrc_class = (GstAudioBaseSrcClass *) klass;
+
+  gobject_class->dispose = gst_jack_audio_src_dispose;
+  gobject_class->set_property = gst_jack_audio_src_set_property;
+  gobject_class->get_property = gst_jack_audio_src_get_property;
+
+  g_object_class_install_property (gobject_class, PROP_CONNECT,
+      g_param_spec_enum ("connect", "Connect",
+          "Specify how the input ports will be connected",
+          GST_TYPE_JACK_CONNECT, DEFAULT_PROP_CONNECT,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_SERVER,
+      g_param_spec_string ("server", "Server",
+          "The Jack server to connect to (NULL = default)",
+          DEFAULT_PROP_SERVER, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstJackAudioSrc:client-name:
+   *
+   * The client name to use.
+   */
+  g_object_class_install_property (gobject_class, PROP_CLIENT_NAME,
+      g_param_spec_string ("client-name", "Client name",
+          "The client name of the Jack instance (NULL = default)",
+          DEFAULT_PROP_CLIENT_NAME,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (gobject_class, PROP_CLIENT,
+      g_param_spec_boxed ("client", "JackClient", "Handle for jack client",
+          GST_TYPE_JACK_CLIENT,
+          GST_PARAM_MUTABLE_READY | G_PARAM_READWRITE |
+          G_PARAM_STATIC_STRINGS));
+
+  /**
+   * GstJackAudioSink:transport:
+   *
+   * The jack transport behaviour for the client.
+   */
+  g_object_class_install_property (gobject_class, PROP_TRANSPORT,
+      g_param_spec_flags ("transport", "Transport mode",
+          "Jack transport behaviour of the client",
+          GST_TYPE_JACK_TRANSPORT, DEFAULT_PROP_TRANSPORT,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  gst_element_class_add_pad_template (gstelement_class,
+      gst_static_pad_template_get (&src_factory));
+
+  gst_element_class_set_static_metadata (gstelement_class,
+      "Audio Source (Jack)", "Source/Audio",
+      "Captures audio from a JACK server",
+      "Tristan Matthews <tristan@sat.qc.ca>");
+
+  gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_jack_audio_src_getcaps);
+  gstaudiobasesrc_class->create_ringbuffer =
+      GST_DEBUG_FUNCPTR (gst_jack_audio_src_create_ringbuffer);
+
+  /* ref class from a thread-safe context to work around missing bit of
+   * thread-safety in GObject */
+  g_type_class_ref (GST_TYPE_JACK_RING_BUFFER);
+
+  gst_jack_audio_client_init ();
+}
+
+static void
+gst_jack_audio_src_init (GstJackAudioSrc * src)
+{
+  //gst_base_src_set_live(GST_BASE_SRC (src), TRUE);
+  src->connect = DEFAULT_PROP_CONNECT;
+  src->server = g_strdup (DEFAULT_PROP_SERVER);
+  src->jclient = NULL;
+  src->ports = NULL;
+  src->port_count = 0;
+  src->buffers = NULL;
+  src->client_name = g_strdup (DEFAULT_PROP_CLIENT_NAME);
+  src->transport = DEFAULT_PROP_TRANSPORT;
+}
+
+static void
+gst_jack_audio_src_dispose (GObject * object)
+{
+  GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
+
+  gst_caps_replace (&src->caps, NULL);
+
+  if (src->client_name != NULL) {
+    g_free (src->client_name);
+    src->client_name = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static void
+gst_jack_audio_src_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
+
+  switch (prop_id) {
+    case PROP_CLIENT_NAME:
+      g_free (src->client_name);
+      src->client_name = g_value_dup_string (value);
+      break;
+    case PROP_CONNECT:
+      src->connect = g_value_get_enum (value);
+      break;
+    case PROP_SERVER:
+      g_free (src->server);
+      src->server = g_value_dup_string (value);
+      break;
+    case PROP_CLIENT:
+      if (GST_STATE (src) == GST_STATE_NULL ||
+          GST_STATE (src) == GST_STATE_READY) {
+        src->jclient = g_value_get_boxed (value);
+      }
+      break;
+    case PROP_TRANSPORT:
+      src->transport = g_value_get_flags (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_jack_audio_src_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (object);
+
+  switch (prop_id) {
+    case PROP_CLIENT_NAME:
+      g_value_set_string (value, src->client_name);
+      break;
+    case PROP_CONNECT:
+      g_value_set_enum (value, src->connect);
+      break;
+    case PROP_SERVER:
+      g_value_set_string (value, src->server);
+      break;
+    case PROP_CLIENT:
+      g_value_set_boxed (value, src->jclient);
+      break;
+    case PROP_TRANSPORT:
+      g_value_set_flags (value, src->transport);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static GstCaps *
+gst_jack_audio_src_getcaps (GstBaseSrc * bsrc, GstCaps * filter)
+{
+  GstJackAudioSrc *src = GST_JACK_AUDIO_SRC (bsrc);
+  const char **ports;
+  gint min, max;
+  gint rate;
+  jack_client_t *client;
+
+  if (src->client == NULL)
+    goto no_client;
+
+  client = gst_jack_audio_client_get_client (src->client);
+
+  if (src->connect == GST_JACK_CONNECT_AUTO) {
+    /* get a port count, this is the number of channels we can automatically
+     * connect. */
+    ports = jack_get_ports (client, NULL, NULL,
+        JackPortIsPhysical | JackPortIsOutput);
+    max = 0;
+    if (ports != NULL) {
+      for (; ports[max]; max++);
+
+      free (ports);
+    } else
+      max = 0;
+  } else {
+    /* we allow any number of pads, something else is going to connect the
+     * pads. */
+    max = G_MAXINT;
+  }
+  min = MIN (1, max);
+
+  rate = jack_get_sample_rate (client);
+
+  GST_DEBUG_OBJECT (src, "got %d-%d ports, samplerate: %d", min, max, rate);
+
+  if (!src->caps) {
+    src->caps = gst_caps_new_simple ("audio/x-raw",
+        "format", G_TYPE_STRING, GST_JACK_FORMAT_STR,
+        "layout", G_TYPE_STRING, "interleaved",
+        "rate", G_TYPE_INT, rate,
+        "channels", GST_TYPE_INT_RANGE, min, max, NULL);
+  }
+  GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, src->caps);
+
+  return gst_caps_ref (src->caps);
+
+  /* ERRORS */
+no_client:
+  {
+    GST_DEBUG_OBJECT (src, "device not open, using template caps");
+    /* base class will get template caps for us when we return NULL */
+    return NULL;
+  }
+}
+
+static GstAudioRingBuffer *
+gst_jack_audio_src_create_ringbuffer (GstAudioBaseSrc * src)
+{
+  GstAudioRingBuffer *buffer;
+
+  buffer = g_object_new (GST_TYPE_JACK_RING_BUFFER, NULL);
+  GST_DEBUG_OBJECT (src, "created ringbuffer @%p", buffer);
+
+  return buffer;
+}
diff --git a/ext/jack/gstjackaudiosrc.h b/ext/jack/gstjackaudiosrc.h
new file mode 100644
index 0000000..97c2891
--- /dev/null
+++ b/ext/jack/gstjackaudiosrc.h
@@ -0,0 +1,99 @@
+/* GStreamer
+ * Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a
+ * copy of this software and associated documentation files (the "Software"),
+ * to deal in the Software without restriction, including without limitation
+ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
+ * and/or sell copies of the Software, and to permit persons to whom the
+ * Software is furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
+ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
+ * DEALINGS IN THE SOFTWARE.
+ *
+ * Alternatively, the contents of this file may be used under the
+ * GNU Lesser General Public License Version 2.1 (the "LGPL"), in
+ * which case the following provisions apply instead of the ones
+ * mentioned above:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_JACK_AUDIO_SRC_H__
+#define __GST_JACK_AUDIO_SRC_H__
+
+#include <jack/jack.h>
+
+#include <gst/gst.h>
+#include <gst/audio/gstaudiosrc.h>
+
+#include "gstjackaudioclient.h"
+#include "gstjack.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_JACK_AUDIO_SRC             (gst_jack_audio_src_get_type())
+#define GST_JACK_AUDIO_SRC(obj)             (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrc))
+#define GST_JACK_AUDIO_SRC_CLASS(klass)     (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass))
+#define GST_JACK_AUDIO_SRC_GET_CLASS(obj)   (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_JACK_AUDIO_SRC,GstJackAudioSrcClass))
+#define GST_IS_JACK_AUDIO_SRC(obj)          (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_AUDIO_SRC))
+#define GST_IS_JACK_AUDIO_SRC_CLASS(klass)  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_AUDIO_SRC))
+
+typedef struct _GstJackAudioSrc GstJackAudioSrc;
+typedef struct _GstJackAudioSrcClass GstJackAudioSrcClass;
+
+struct _GstJackAudioSrc
+{
+    GstAudioBaseSrc src;
+
+    /*< private >*/
+    /* cached caps */
+    GstCaps         *caps;
+
+    /* properties */
+    GstJackConnect   connect;
+    gchar           *server;
+    jack_client_t   *jclient;
+    gchar           *client_name;
+    guint            transport;
+
+    /* our client */
+    GstJackAudioClient *client;
+
+    /* our ports */
+    jack_port_t    **ports;
+    int port_count;
+    sample_t **buffers;
+};
+
+struct _GstJackAudioSrcClass
+{
+    GstAudioBaseSrcClass parent_class;
+};
+
+GType gst_jack_audio_src_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_JACK_AUDIO_SRC_H__ */
diff --git a/ext/jack/gstjackringbuffer.h b/ext/jack/gstjackringbuffer.h
new file mode 100644
index 0000000..94de4b8
--- /dev/null
+++ b/ext/jack/gstjackringbuffer.h
@@ -0,0 +1,88 @@
+/*
+ * GStreamer
+ * Copyright (C) 2006 Wim Taymans <wim@fluendo.com>
+ * Copyright (C) 2008 Tristan Matthews <tristan@sat.qc.ca>
+ *
+ * Permission is hereby granted, free of charge, to any person obtaining a
+ * copy of this software and associated documentation files (the "Software"),
+ * to deal in the Software without restriction, including without limitation
+ * the rights to use, copy, modify, merge, publish, distribute, sublicense,
+ * and/or sell copies of the Software, and to permit persons to whom the
+ * Software is furnished to do so, subject to the following conditions:
+ *
+ * The above copyright notice and this permission notice shall be included in
+ * all copies or substantial portions of the Software.
+ *
+ * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
+ * IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
+ * FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE
+ * AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
+ * LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
+ * FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
+ * DEALINGS IN THE SOFTWARE.
+ *
+ * Alternatively, the contents of this file may be used under the
+ * GNU Lesser General Public License Version 2.1 (the "LGPL"), in
+ * which case the following provisions apply instead of the ones
+ * mentioned above:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef __GST_JACK_RING_BUFFER_H__
+#define __GST_JACK_RING_BUFFER_H__
+
+#define GST_TYPE_JACK_RING_BUFFER               (gst_jack_ring_buffer_get_type())
+#define GST_JACK_RING_BUFFER(obj)               (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_JACK_RING_BUFFER,GstJackRingBuffer))
+#define GST_JACK_RING_BUFFER_CLASS(klass)       (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass))
+#define GST_JACK_RING_BUFFER_GET_CLASS(obj)     (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_JACK_RING_BUFFER,GstJackRingBufferClass))
+#define GST_JACK_RING_BUFFER_CAST(obj)          ((GstJackRingBuffer *)obj)
+#define GST_IS_JACK_RING_BUFFER(obj)            (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_JACK_RING_BUFFER))
+#define GST_IS_JACK_RING_BUFFER_CLASS(klass)    (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_JACK_RING_BUFFER))
+
+typedef struct _GstJackRingBuffer GstJackRingBuffer;
+typedef struct _GstJackRingBufferClass GstJackRingBufferClass;
+
+struct _GstJackRingBuffer
+{
+  GstAudioRingBuffer object;
+
+  gint sample_rate;
+  gint buffer_size;
+  gint channels;
+};
+
+struct _GstJackRingBufferClass
+{
+  GstAudioRingBufferClass parent_class;
+};
+
+static void gst_jack_ring_buffer_class_init(GstJackRingBufferClass * klass);
+static void gst_jack_ring_buffer_init(GstJackRingBuffer * ringbuffer,
+    GstJackRingBufferClass * klass);
+
+static GstAudioRingBufferClass *ring_parent_class = NULL;
+
+static gboolean gst_jack_ring_buffer_open_device(GstAudioRingBuffer * buf);
+static gboolean gst_jack_ring_buffer_close_device(GstAudioRingBuffer * buf);
+static gboolean gst_jack_ring_buffer_acquire(GstAudioRingBuffer * buf,GstAudioRingBufferSpec * spec);
+static gboolean gst_jack_ring_buffer_release(GstAudioRingBuffer * buf);
+static gboolean gst_jack_ring_buffer_start(GstAudioRingBuffer * buf);
+static gboolean gst_jack_ring_buffer_pause(GstAudioRingBuffer * buf);
+static gboolean gst_jack_ring_buffer_stop(GstAudioRingBuffer * buf);
+static guint gst_jack_ring_buffer_delay(GstAudioRingBuffer * buf);
+
+#endif 
diff --git a/ext/jack/gstjackutil.c b/ext/jack/gstjackutil.c
new file mode 100644
index 0000000..490e14a
--- /dev/null
+++ b/ext/jack/gstjackutil.c
@@ -0,0 +1,110 @@
+/* GStreamer Jack utility functions
+ * Copyright (C) 2010 Tristan Matthews <tristan@sat.qc.ca>
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#include "gstjackutil.h"
+#include <gst/audio/audio.h>
+
+static const GstAudioChannelPosition default_positions[8][8] = {
+  /* 1 channel */
+  {
+        GST_AUDIO_CHANNEL_POSITION_MONO,
+      },
+  /* 2 channels */
+  {
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+      },
+  /* 3 channels (2.1) */
+  {
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_LFE1,        /* or FRONT_CENTER for 3.0? */
+      },
+  /* 4 channels (4.0 or 3.1?) */
+  {
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+      },
+  /* 5 channels */
+  {
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+      },
+  /* 6 channels */
+  {
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_LFE1,
+      },
+  /* 7 channels */
+  {
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_LFE1,
+        GST_AUDIO_CHANNEL_POSITION_REAR_CENTER,
+      },
+  /* 8 channels */
+  {
+        GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_REAR_RIGHT,
+        GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
+        GST_AUDIO_CHANNEL_POSITION_LFE1,
+        GST_AUDIO_CHANNEL_POSITION_SIDE_LEFT,
+        GST_AUDIO_CHANNEL_POSITION_SIDE_RIGHT,
+      }
+};
+
+
+/* if channels are less than or equal to 8, we set a default layout,
+ * otherwise set layout to an array of GST_AUDIO_CHANNEL_POSITION_NONE */
+void
+gst_jack_set_layout (GstAudioRingBuffer * buffer, GstAudioRingBufferSpec * spec)
+{
+  gint i;
+
+  if (spec->info.channels <= 8) {
+    for (i = 0; i < spec->info.channels; i++)
+      spec->info.position[i] = default_positions[spec->info.channels - 1][i];
+    gst_audio_channel_positions_to_valid_order (spec->info.position,
+        spec->info.channels);
+    gst_audio_ring_buffer_set_channel_positions (buffer,
+        default_positions[spec->info.channels - 1]);
+  } else {
+    spec->info.flags |= GST_AUDIO_FLAG_UNPOSITIONED;
+    for (i = 0; i < G_N_ELEMENTS (spec->info.position); i++)
+      spec->info.position[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
+    gst_audio_ring_buffer_set_channel_positions (buffer, spec->info.position);
+  }
+
+  gst_caps_unref (spec->caps);
+  spec->caps = gst_audio_info_to_caps (&spec->info);
+}
diff --git a/ext/jack/gstjackutil.h b/ext/jack/gstjackutil.h
new file mode 100644
index 0000000..7a4bcc5
--- /dev/null
+++ b/ext/jack/gstjackutil.h
@@ -0,0 +1,31 @@
+/* GStreamer
+ * Copyright (C) 2010 Tristan Matthews <tristan@sat.qc.ca>
+ *
+ * gstjackutil.h:
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Library General Public
+ * License as published by the Free Software Foundation; either
+ * version 2 of the License, or (at your option) any later version.
+ *
+ * This library 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
+ * Library General Public License for more details.
+ *
+ * You should have received a copy of the GNU Library General Public
+ * License along with this library; if not, write to the
+ * Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
+ * Boston, MA 02110-1301, USA.
+ */
+
+#ifndef _GST_JACK_UTIL_H_
+#define _GST_JACK_UTIL_H_
+
+#include <gst/gst.h>
+#include <gst/audio/audio.h>
+
+void
+gst_jack_set_layout (GstAudioRingBuffer * buffer, GstAudioRingBufferSpec *spec);
+
+#endif  // _GST_JACK_UTIL_H_