tizen 2.4 releasetizen_2.4_mobile_releasesubmit/tizen_2.4/20151028.064339accepted/tizen/2.4/mobile/20151029.032704accepted/tizen_2.4_mobile

12 files changed, 3814 insertions, 0 deletions

diff --git a/gst/replaygain/Makefile.am b/gst/replaygain/Makefile.am
new file mode 100644
index 0000000..d352f18
--- /dev/null
+++ b/gst/replaygain/Makefile.am
@@ -0,0 +1,38 @@
+plugin_LTLIBRARIES = libgstreplaygain.la
+
+libgstreplaygain_la_SOURCES = \
+	gstrganalysis.c       \
+	gstrglimiter.c        \
+	gstrgvolume.c         \
+	replaygain.c          \
+	rganalysis.c
+libgstreplaygain_la_CFLAGS  = \
+	$(GST_PLUGINS_BASE_CFLAGS) $(GST_BASE_CFLAGS) $(GST_CFLAGS)
+libgstreplaygain_la_LIBADD  = \
+	$(GST_PLUGINS_BASE_LIBS) -lgstpbutils-$(GST_API_VERSION) -lgstaudio-$(GST_API_VERSION)\
+	$(GST_BASE_LIBS) $(GST_LIBS) $(LIBM)
+libgstreplaygain_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+libgstreplaygain_la_LIBTOOLFLAGS = $(GST_PLUGIN_LIBTOOLFLAGS)
+
+# headers we need but don't want installed
+noinst_HEADERS =         \
+	gstrganalysis.h  \
+	gstrglimiter.h   \
+	gstrgvolume.h    \
+	replaygain.h     \
+	rganalysis.h
+
+
+Android.mk: Makefile.am $(BUILT_SOURCES)
+	androgenizer \
+	-:PROJECT libgstreplaygain -:SHARED libgstreplaygain \
+	 -:TAGS eng debug \
+         -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
+	 -:SOURCES $(libgstreplaygain_la_SOURCES) \
+	 -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstreplaygain_la_CFLAGS) \
+	 -:LDFLAGS $(libgstreplaygain_la_LDFLAGS) \
+	           $(libgstreplaygain_la_LIBADD) \
+	           -ldl \
+	 -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
+		       LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
+	> $@
diff --git a/gst/replaygain/Makefile.in b/gst/replaygain/Makefile.in
new file mode 100755
index 0000000..11f7d05
--- /dev/null
+++ b/gst/replaygain/Makefile.in
@@ -0,0 +1,961 @@
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+	     $$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
+
+
+Android.mk: Makefile.am $(BUILT_SOURCES)
+	androgenizer \
+	-:PROJECT libgstreplaygain -:SHARED libgstreplaygain \
+	 -:TAGS eng debug \
+         -:REL_TOP $(top_srcdir) -:ABS_TOP $(abs_top_srcdir) \
+	 -:SOURCES $(libgstreplaygain_la_SOURCES) \
+	 -:CFLAGS $(DEFS) $(DEFAULT_INCLUDES) $(libgstreplaygain_la_CFLAGS) \
+	 -:LDFLAGS $(libgstreplaygain_la_LDFLAGS) \
+	           $(libgstreplaygain_la_LIBADD) \
+	           -ldl \
+	 -:PASSTHROUGH LOCAL_ARM_MODE:=arm \
+		       LOCAL_MODULE_PATH:='$$(TARGET_OUT)/lib/gstreamer-0.10' \
+	> $@
+
+# 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/gst/replaygain/gstrganalysis.c b/gst/replaygain/gstrganalysis.c
new file mode 100755
index 0000000..98c3ca6
--- /dev/null
+++ b/gst/replaygain/gstrganalysis.c
@@ -0,0 +1,711 @@
+/* GStreamer ReplayGain analysis
+ *
+ * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
+ * 
+ * gstrganalysis.c: Element that performs the ReplayGain analysis
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+/**
+ * SECTION:element-rganalysis
+ * @see_also: #GstRgVolume
+ *
+ * This element analyzes raw audio sample data in accordance with the proposed
+ * <ulink url="http://replaygain.org">ReplayGain standard</ulink> for
+ * calculating the ideal replay gain for music tracks and albums.  The element
+ * is designed as a pass-through filter that never modifies any data.  As it
+ * receives an EOS event, it finalizes the ongoing analysis and generates a tag
+ * list containing the results.  It is sent downstream with a tag event and
+ * posted on the message bus with a tag message.  The EOS event is forwarded as
+ * normal afterwards.  Result tag lists at least contain the tags
+ * #GST_TAG_TRACK_GAIN, #GST_TAG_TRACK_PEAK and #GST_TAG_REFERENCE_LEVEL.
+ * 
+ * Because the generated metadata tags become available at the end of streams,
+ * downstream muxer and encoder elements are normally unable to save them in
+ * their output since they generally save metadata in the file header.
+ * Therefore, it is often necessary that applications read the results in a bus
+ * event handler for the tag message.  Obtaining the values this way is always
+ * needed for album processing (see #GstRgAnalysis:num-tracks property) since
+ * the album gain and peak values need to be associated with all tracks of an
+ * album, not just the last one.
+ * 
+ * <refsect2>
+ * <title>Example launch lines</title>
+ * |[
+ * gst-launch-1.0 -t audiotestsrc wave=sine num-buffers=512 ! rganalysis ! fakesink
+ * ]| Analyze a simple test waveform
+ * |[
+ * gst-launch-1.0 -t filesrc location=filename.ext ! decodebin \
+ *     ! audioconvert ! audioresample ! rganalysis ! fakesink
+ * ]| Analyze a given file
+ * |[
+ * gst-launch-1.0 -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav \
+ *     ! wavparse ! rganalysis ! fakesink
+ * ]| Analyze the pink noise reference file
+ * <para>
+ * The above launch line yields a result gain of +6 dB (instead of the expected
+ * +0 dB).  This is not in error, refer to the #GstRgAnalysis:reference-level
+ * property documentation for more information.
+ * </para>
+ * </refsect2>
+ * <refsect2>
+ * <title>Acknowledgements</title>
+ * <para>
+ * This element is based on code used in the <ulink
+ * url="http://sjeng.org/vorbisgain.html">vorbisgain</ulink> program and many
+ * others.  The relevant parts are copyrighted by David Robinson, Glen Sawyer
+ * and Frank Klemm.
+ * </para>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+#include <gst/audio/audio.h>
+
+#include "gstrganalysis.h"
+#include "replaygain.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_rg_analysis_debug);
+#define GST_CAT_DEFAULT gst_rg_analysis_debug
+
+/* Default property value. */
+#define FORCED_DEFAULT TRUE
+#define DEFAULT_MESSAGE FALSE
+
+enum
+{
+  PROP_0,
+  PROP_NUM_TRACKS,
+  PROP_FORCED,
+  PROP_REFERENCE_LEVEL,
+  PROP_MESSAGE
+};
+
+/* The ReplayGain algorithm is intended for use with mono and stereo
+ * audio.  The used implementation has filter coefficients for the
+ * "usual" sample rates in the 8000 to 48000 Hz range. */
+#define REPLAY_GAIN_CAPS "audio/x-raw," \
+  "format = (string) { "GST_AUDIO_NE(F32)","GST_AUDIO_NE(S16)" }, "     \
+  "layout = (string) interleaved, "                                     \
+  "channels = (int) 1, "                                                \
+  "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, "     \
+  "44100, 48000 }; "                                                    \
+  "audio/x-raw,"                                                        \
+  "format = (string) { "GST_AUDIO_NE(F32)","GST_AUDIO_NE(S16)" }, "     \
+  "layout = (string) interleaved, "                                     \
+  "channels = (int) 2, "                                                \
+  "channel-mask = (bitmask) 0x3, "                                      \
+  "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, "     \
+  "44100, 48000 }"
+
+static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (REPLAY_GAIN_CAPS));
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (REPLAY_GAIN_CAPS));
+
+#define gst_rg_analysis_parent_class parent_class
+G_DEFINE_TYPE (GstRgAnalysis, gst_rg_analysis, GST_TYPE_BASE_TRANSFORM);
+
+static void gst_rg_analysis_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rg_analysis_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static gboolean gst_rg_analysis_start (GstBaseTransform * base);
+static gboolean gst_rg_analysis_set_caps (GstBaseTransform * base,
+    GstCaps * incaps, GstCaps * outcaps);
+static GstFlowReturn gst_rg_analysis_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf);
+static gboolean gst_rg_analysis_sink_event (GstBaseTransform * base,
+    GstEvent * event);
+static gboolean gst_rg_analysis_stop (GstBaseTransform * base);
+
+static void gst_rg_analysis_handle_tags (GstRgAnalysis * filter,
+    const GstTagList * tag_list);
+static void gst_rg_analysis_handle_eos (GstRgAnalysis * filter);
+static gboolean gst_rg_analysis_track_result (GstRgAnalysis * filter,
+    GstTagList ** tag_list);
+static gboolean gst_rg_analysis_album_result (GstRgAnalysis * filter,
+    GstTagList ** tag_list);
+
+static void
+gst_rg_analysis_class_init (GstRgAnalysisClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *element_class;
+  GstBaseTransformClass *trans_class;
+
+  gobject_class = (GObjectClass *) klass;
+  element_class = (GstElementClass *) klass;
+
+  gobject_class->set_property = gst_rg_analysis_set_property;
+  gobject_class->get_property = gst_rg_analysis_get_property;
+
+  /**
+   * GstRgAnalysis:num-tracks:
+   *
+   * Number of remaining album tracks.
+   * 
+   * Analyzing several streams sequentially and assigning them a common result
+   * gain is known as "album processing".  If this gain is used during playback
+   * (by switching to "album mode"), all tracks of an album receive the same
+   * amplification.  This keeps the relative volume levels between the tracks
+   * intact.  To enable this, set this property to the number of streams that
+   * will be processed as album tracks.
+   *
+   * Every time an EOS event is received, the value of this property is
+   * decremented by one.  As it reaches zero, it is assumed that the last track
+   * of the album finished.  The tag list for the final stream will contain the
+   * additional tags #GST_TAG_ALBUM_GAIN and #GST_TAG_ALBUM_PEAK.  All other
+   * streams just get the two track tags posted because the values for the album
+   * tags are not known before all tracks are analyzed.  Applications need to
+   * ensure that the album gain and peak values are also associated with the
+   * other tracks when storing the results.
+   *
+   * If the total number of album tracks is unknown beforehand, just ensure that
+   * the value is greater than 1 before each track starts.  Then before the end
+   * of the last track, set it to the value 1.
+   *
+   * To perform album processing, the element has to preserve data between
+   * streams.  This cannot survive a state change to the NULL or READY state.
+   * If you change your pipeline's state to NULL or READY between tracks, lock
+   * the element's state using gst_element_set_locked_state() when it is in
+   * PAUSED or PLAYING.
+   */
+  g_object_class_install_property (gobject_class, PROP_NUM_TRACKS,
+      g_param_spec_int ("num-tracks", "Number of album tracks",
+          "Number of remaining album tracks", 0, G_MAXINT, 0,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRgAnalysis:forced:
+   *
+   * Whether to analyze streams even when ReplayGain tags exist.
+   *
+   * For assisting transcoder/converter applications, the element can silently
+   * skip the processing of streams that already contain the necessary tags.
+   * Data will flow as usual but the element will not consume CPU time and will
+   * not generate result tags.  To enable possible skipping, set this property
+   * to #FALSE.
+   *
+   * If used in conjunction with <link linkend="GstRgAnalysis--num-tracks">album
+   * processing</link>, the element will skip the number of remaining album
+   * tracks if a full set of tags is found for the first track.  If a subsequent
+   * track of the album is missing tags, processing cannot start again.  If this
+   * is undesired, the application has to scan all files beforehand and enable
+   * forcing of processing if needed.
+   */
+  g_object_class_install_property (gobject_class, PROP_FORCED,
+      g_param_spec_boolean ("forced", "Forced",
+          "Analyze even if ReplayGain tags exist",
+          FORCED_DEFAULT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRgAnalysis:reference-level:
+   *
+   * Reference level [dB].
+   *
+   * Analyzing the ReplayGain pink noise reference waveform computes a result of
+   * +6 dB instead of the expected 0 dB.  This is because the default reference
+   * level is 89 dB.  To obtain values as lined out in the original proposal of
+   * ReplayGain, set this property to 83.
+   *
+   * Almost all software uses 89 dB as a reference however, and this value has
+   * become the new official value.  That is to say, while the change has been
+   * acclaimed by the author of the ReplayGain proposal, the <ulink
+   * url="http://replaygain.org">webpage</ulink> is still outdated at the time
+   * of this writing.
+   *
+   * The value was changed because the original proposal recommends a default
+   * pre-amp value of +6 dB for playback.  This seemed a bit odd, as it means
+   * that the algorithm has the general tendency to produce adjustment values
+   * that are 6 dB too low.  Bumping the reference level by 6 dB compensated for
+   * this.
+   *
+   * The problem of the reference level being ambiguous for lack of concise
+   * standardization is to be solved by adopting the #GST_TAG_REFERENCE_LEVEL
+   * tag, which allows to store the used value alongside the gain values.
+   */
+  g_object_class_install_property (gobject_class, PROP_REFERENCE_LEVEL,
+      g_param_spec_double ("reference-level", "Reference level",
+          "Reference level [dB]", 0.0, 150., RG_REFERENCE_LEVEL,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MESSAGE,
+      g_param_spec_boolean ("message", "Message",
+          "Post statics messages",
+          DEFAULT_MESSAGE,
+          G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
+
+  trans_class = (GstBaseTransformClass *) klass;
+  trans_class->start = GST_DEBUG_FUNCPTR (gst_rg_analysis_start);
+  trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_rg_analysis_set_caps);
+  trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_analysis_transform_ip);
+  trans_class->sink_event = GST_DEBUG_FUNCPTR (gst_rg_analysis_sink_event);
+  trans_class->stop = GST_DEBUG_FUNCPTR (gst_rg_analysis_stop);
+  trans_class->passthrough_on_same_caps = TRUE;
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_factory));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_factory));
+  gst_element_class_set_static_metadata (element_class, "ReplayGain analysis",
+      "Filter/Analyzer/Audio",
+      "Perform the ReplayGain analysis",
+      "Ren\xc3\xa9 Stadler <mail@renestadler.de>");
+
+  GST_DEBUG_CATEGORY_INIT (gst_rg_analysis_debug, "rganalysis", 0,
+      "ReplayGain analysis element");
+}
+
+static void
+gst_rg_analysis_init (GstRgAnalysis * filter)
+{
+  GstBaseTransform *base = GST_BASE_TRANSFORM (filter);
+
+  gst_base_transform_set_gap_aware (base, TRUE);
+
+  filter->num_tracks = 0;
+  filter->forced = FORCED_DEFAULT;
+  filter->message = DEFAULT_MESSAGE;
+  filter->reference_level = RG_REFERENCE_LEVEL;
+
+  filter->ctx = NULL;
+  filter->analyze = NULL;
+}
+
+static void
+gst_rg_analysis_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRgAnalysis *filter = GST_RG_ANALYSIS (object);
+
+  GST_OBJECT_LOCK (filter);
+  switch (prop_id) {
+    case PROP_NUM_TRACKS:
+      filter->num_tracks = g_value_get_int (value);
+      break;
+    case PROP_FORCED:
+      filter->forced = g_value_get_boolean (value);
+      break;
+    case PROP_REFERENCE_LEVEL:
+      filter->reference_level = g_value_get_double (value);
+      break;
+    case PROP_MESSAGE:
+      filter->message = g_value_get_boolean (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+  GST_OBJECT_UNLOCK (filter);
+}
+
+static void
+gst_rg_analysis_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRgAnalysis *filter = GST_RG_ANALYSIS (object);
+
+  GST_OBJECT_LOCK (filter);
+  switch (prop_id) {
+    case PROP_NUM_TRACKS:
+      g_value_set_int (value, filter->num_tracks);
+      break;
+    case PROP_FORCED:
+      g_value_set_boolean (value, filter->forced);
+      break;
+    case PROP_REFERENCE_LEVEL:
+      g_value_set_double (value, filter->reference_level);
+      break;
+    case PROP_MESSAGE:
+      g_value_set_boolean (value, filter->message);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+  GST_OBJECT_UNLOCK (filter);
+}
+
+static void
+gst_rg_analysis_post_message (gpointer rganalysis, GstClockTime timestamp,
+    GstClockTime duration, gdouble rglevel)
+{
+  GstRgAnalysis *filter = GST_RG_ANALYSIS (rganalysis);
+  if (filter->message) {
+    GstMessage *m;
+
+    m = gst_message_new_element (GST_OBJECT_CAST (rganalysis),
+        gst_structure_new ("rganalysis",
+            "timestamp", G_TYPE_UINT64, timestamp,
+            "duration", G_TYPE_UINT64, duration,
+            "rglevel", G_TYPE_DOUBLE, rglevel, NULL));
+
+    gst_element_post_message (GST_ELEMENT_CAST (rganalysis), m);
+  }
+}
+
+
+static gboolean
+gst_rg_analysis_start (GstBaseTransform * base)
+{
+  GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
+
+  filter->ignore_tags = FALSE;
+  filter->skip = FALSE;
+  filter->has_track_gain = FALSE;
+  filter->has_track_peak = FALSE;
+  filter->has_album_gain = FALSE;
+  filter->has_album_peak = FALSE;
+
+  filter->ctx = rg_analysis_new ();
+  GST_OBJECT_LOCK (filter);
+  rg_analysis_init_silence_detection (filter->ctx, gst_rg_analysis_post_message,
+      filter);
+  GST_OBJECT_UNLOCK (filter);
+  filter->analyze = NULL;
+
+  GST_LOG_OBJECT (filter, "started");
+
+  return TRUE;
+}
+
+static gboolean
+gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
+    GstCaps * out_caps)
+{
+  GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
+  GstAudioInfo info;
+  gint rate, channels;
+
+  g_return_val_if_fail (filter->ctx != NULL, FALSE);
+
+  GST_DEBUG_OBJECT (filter,
+      "set_caps in %" GST_PTR_FORMAT " out %" GST_PTR_FORMAT,
+      in_caps, out_caps);
+
+  if (!gst_audio_info_from_caps (&info, in_caps))
+    goto invalid_format;
+
+  rate = GST_AUDIO_INFO_RATE (&info);
+
+  if (!rg_analysis_set_sample_rate (filter->ctx, rate))
+    goto invalid_format;
+
+  channels = GST_AUDIO_INFO_CHANNELS (&info);
+
+  if (channels < 1 || channels > 2)
+    goto invalid_format;
+
+  switch (GST_AUDIO_INFO_FORMAT (&info)) {
+    case GST_AUDIO_FORMAT_F32:
+      /* The depth is not variable for float formats of course.  It just
+       * makes the transform function nice and simple if the
+       * rg_analysis_analyze_* functions have a common signature. */
+      filter->depth = sizeof (gfloat) * 8;
+
+      if (channels == 1)
+        filter->analyze = rg_analysis_analyze_mono_float;
+      else
+        filter->analyze = rg_analysis_analyze_stereo_float;
+
+      break;
+    case GST_AUDIO_FORMAT_S16:
+      filter->depth = sizeof (gint16) * 8;
+
+      if (channels == 1)
+        filter->analyze = rg_analysis_analyze_mono_int16;
+      else
+        filter->analyze = rg_analysis_analyze_stereo_int16;
+      break;
+    default:
+      goto invalid_format;
+  }
+
+  return TRUE;
+
+  /* Errors. */
+invalid_format:
+  {
+    filter->analyze = NULL;
+    GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
+        ("Invalid incoming caps: %" GST_PTR_FORMAT, in_caps), (NULL));
+    return FALSE;
+  }
+}
+
+static GstFlowReturn
+gst_rg_analysis_transform_ip (GstBaseTransform * base, GstBuffer * buf)
+{
+  GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
+  GstMapInfo map;
+
+  g_return_val_if_fail (filter->ctx != NULL, GST_FLOW_FLUSHING);
+  g_return_val_if_fail (filter->analyze != NULL, GST_FLOW_NOT_NEGOTIATED);
+
+  if (filter->skip)
+    return GST_FLOW_OK;
+
+  gst_buffer_map (buf, &map, GST_MAP_READ);
+  GST_LOG_OBJECT (filter, "processing buffer of size %" G_GSIZE_FORMAT,
+      map.size);
+
+  rg_analysis_start_buffer (filter->ctx, GST_BUFFER_TIMESTAMP (buf));
+  filter->analyze (filter->ctx, map.data, map.size, filter->depth);
+
+  gst_buffer_unmap (buf, &map);
+
+  return GST_FLOW_OK;
+}
+
+static gboolean
+gst_rg_analysis_sink_event (GstBaseTransform * base, GstEvent * event)
+{
+  GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
+
+  g_return_val_if_fail (filter->ctx != NULL, TRUE);
+
+  switch (GST_EVENT_TYPE (event)) {
+
+    case GST_EVENT_EOS:
+    {
+      GST_LOG_OBJECT (filter, "received EOS event");
+
+      gst_rg_analysis_handle_eos (filter);
+
+      GST_LOG_OBJECT (filter, "passing on EOS event");
+
+      break;
+    }
+    case GST_EVENT_TAG:
+    {
+      GstTagList *tag_list;
+
+      /* The reference to the tag list is borrowed. */
+      gst_event_parse_tag (event, &tag_list);
+      gst_rg_analysis_handle_tags (filter, tag_list);
+
+      break;
+    }
+    default:
+      break;
+  }
+
+  return GST_BASE_TRANSFORM_CLASS (parent_class)->sink_event (base, event);
+}
+
+static gboolean
+gst_rg_analysis_stop (GstBaseTransform * base)
+{
+  GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
+
+  g_return_val_if_fail (filter->ctx != NULL, FALSE);
+
+  rg_analysis_destroy (filter->ctx);
+  filter->ctx = NULL;
+
+  GST_LOG_OBJECT (filter, "stopped");
+
+  return TRUE;
+}
+
+/* FIXME: handle global vs. stream-tags? */
+static void
+gst_rg_analysis_handle_tags (GstRgAnalysis * filter,
+    const GstTagList * tag_list)
+{
+  gboolean album_processing = (filter->num_tracks > 0);
+  gdouble dummy;
+
+  if (!album_processing)
+    filter->ignore_tags = FALSE;
+
+  if (filter->skip && album_processing) {
+    GST_DEBUG_OBJECT (filter, "ignoring tag event: skipping album");
+    return;
+  } else if (filter->skip) {
+    GST_DEBUG_OBJECT (filter, "ignoring tag event: skipping track");
+    return;
+  } else if (filter->ignore_tags) {
+    GST_DEBUG_OBJECT (filter, "ignoring tag event: cannot skip anyways");
+    return;
+  }
+
+  filter->has_track_gain |= gst_tag_list_get_double (tag_list,
+      GST_TAG_TRACK_GAIN, &dummy);
+  filter->has_track_peak |= gst_tag_list_get_double (tag_list,
+      GST_TAG_TRACK_PEAK, &dummy);
+  filter->has_album_gain |= gst_tag_list_get_double (tag_list,
+      GST_TAG_ALBUM_GAIN, &dummy);
+  filter->has_album_peak |= gst_tag_list_get_double (tag_list,
+      GST_TAG_ALBUM_PEAK, &dummy);
+
+  if (!(filter->has_track_gain && filter->has_track_peak)) {
+    GST_DEBUG_OBJECT (filter, "track tags not complete yet");
+    return;
+  }
+
+  if (album_processing && !(filter->has_album_gain && filter->has_album_peak)) {
+    GST_DEBUG_OBJECT (filter, "album tags not complete yet");
+    return;
+  }
+
+  if (filter->forced) {
+    GST_DEBUG_OBJECT (filter,
+        "existing tags are sufficient, but processing anyway (forced)");
+    return;
+  }
+
+  filter->skip = TRUE;
+  rg_analysis_reset (filter->ctx);
+
+  if (!album_processing) {
+    GST_DEBUG_OBJECT (filter,
+        "existing tags are sufficient, will not process this track");
+  } else {
+    GST_DEBUG_OBJECT (filter,
+        "existing tags are sufficient, will not process this album");
+  }
+}
+
+static void
+gst_rg_analysis_handle_eos (GstRgAnalysis * filter)
+{
+  gboolean album_processing = (filter->num_tracks > 0);
+  gboolean album_finished = (filter->num_tracks == 1);
+  gboolean album_skipping = album_processing && filter->skip;
+
+  filter->has_track_gain = FALSE;
+  filter->has_track_peak = FALSE;
+
+  if (album_finished) {
+    filter->ignore_tags = FALSE;
+    filter->skip = FALSE;
+    filter->has_album_gain = FALSE;
+    filter->has_album_peak = FALSE;
+  } else if (!album_skipping) {
+    filter->skip = FALSE;
+  }
+
+  /* We might have just fully processed a track because it has
+   * incomplete tags.  If we do album processing and allow skipping
+   * (not forced), prevent switching to skipping if a later track with
+   * full tags comes along: */
+  if (!filter->forced && album_processing && !album_finished)
+    filter->ignore_tags = TRUE;
+
+  if (!filter->skip) {
+    GstTagList *tag_list = NULL;
+    gboolean track_success;
+    gboolean album_success = FALSE;
+
+    track_success = gst_rg_analysis_track_result (filter, &tag_list);
+
+    if (album_finished)
+      album_success = gst_rg_analysis_album_result (filter, &tag_list);
+    else if (!album_processing)
+      rg_analysis_reset_album (filter->ctx);
+
+    if (track_success || album_success) {
+      GST_LOG_OBJECT (filter, "posting tag list with results");
+      gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
+          GST_TAG_REFERENCE_LEVEL, filter->reference_level, NULL);
+      /* This takes ownership of our reference to the list */
+      gst_pad_push_event (GST_BASE_TRANSFORM_SRC_PAD (filter),
+          gst_event_new_tag (tag_list));
+      tag_list = NULL;
+    }
+  }
+
+  if (album_processing) {
+    filter->num_tracks--;
+
+    if (!album_finished) {
+      GST_DEBUG_OBJECT (filter, "album not finished yet (num-tracks is now %u)",
+          filter->num_tracks);
+    } else {
+      GST_DEBUG_OBJECT (filter, "album finished (num-tracks is now 0)");
+    }
+  }
+
+  if (album_processing)
+    g_object_notify (G_OBJECT (filter), "num-tracks");
+}
+
+/* FIXME: return tag list (lists?) based on input tags.. */
+static gboolean
+gst_rg_analysis_track_result (GstRgAnalysis * filter, GstTagList ** tag_list)
+{
+  gboolean track_success;
+  gdouble track_gain, track_peak;
+
+  track_success = rg_analysis_track_result (filter->ctx, &track_gain,
+      &track_peak);
+
+  if (track_success) {
+    track_gain += filter->reference_level - RG_REFERENCE_LEVEL;
+    GST_INFO_OBJECT (filter, "track gain is %+.2f dB, peak %.6f", track_gain,
+        track_peak);
+  } else {
+    GST_INFO_OBJECT (filter, "track was too short to analyze");
+  }
+
+  if (track_success) {
+    if (*tag_list == NULL)
+      *tag_list = gst_tag_list_new_empty ();
+    gst_tag_list_add (*tag_list, GST_TAG_MERGE_APPEND,
+        GST_TAG_TRACK_PEAK, track_peak, GST_TAG_TRACK_GAIN, track_gain, NULL);
+  }
+
+  return track_success;
+}
+
+static gboolean
+gst_rg_analysis_album_result (GstRgAnalysis * filter, GstTagList ** tag_list)
+{
+  gboolean album_success;
+  gdouble album_gain, album_peak;
+
+  album_success = rg_analysis_album_result (filter->ctx, &album_gain,
+      &album_peak);
+
+  if (album_success) {
+    album_gain += filter->reference_level - RG_REFERENCE_LEVEL;
+    GST_INFO_OBJECT (filter, "album gain is %+.2f dB, peak %.6f", album_gain,
+        album_peak);
+  } else {
+    GST_INFO_OBJECT (filter, "album was too short to analyze");
+  }
+
+  if (album_success) {
+    if (*tag_list == NULL)
+      *tag_list = gst_tag_list_new_empty ();
+    gst_tag_list_add (*tag_list, GST_TAG_MERGE_APPEND,
+        GST_TAG_ALBUM_PEAK, album_peak, GST_TAG_ALBUM_GAIN, album_gain, NULL);
+  }
+
+  return album_success;
+}
diff --git a/gst/replaygain/gstrganalysis.h b/gst/replaygain/gstrganalysis.h
new file mode 100644
index 0000000..0d68e63
--- /dev/null
+++ b/gst/replaygain/gstrganalysis.h
@@ -0,0 +1,86 @@
+/* GStreamer ReplayGain analysis
+ *
+ * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
+ *
+ * gstrganalysis.h: Element that performs the ReplayGain analysis
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifndef __GST_RG_ANALYSIS_H__
+#define __GST_RG_ANALYSIS_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+
+#include "rganalysis.h"
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RG_ANALYSIS \
+  (gst_rg_analysis_get_type())
+#define GST_RG_ANALYSIS(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_ANALYSIS,GstRgAnalysis))
+#define GST_RG_ANALYSIS_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_ANALYSIS,GstRgAnalysisClass))
+#define GST_IS_RG_ANALYSIS(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_ANALYSIS))
+#define GST_IS_RG_ANALYSIS_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_ANALYSIS))
+typedef struct _GstRgAnalysis GstRgAnalysis;
+typedef struct _GstRgAnalysisClass GstRgAnalysisClass;
+
+/**
+ * GstRgAnalysis:
+ *
+ * Opaque data structure.
+ */
+struct _GstRgAnalysis
+{
+  GstBaseTransform element;
+
+  /*< private >*/
+
+  RgAnalysisCtx *ctx;
+  void (*analyze) (RgAnalysisCtx * ctx, gconstpointer data, gsize size,
+      guint depth);
+  gint depth;
+
+  /* Property values. */
+  guint num_tracks;
+  gdouble reference_level;
+  gboolean forced;
+  gboolean message;
+
+  /* State machinery for skipping. */
+  gboolean ignore_tags;
+  gboolean skip;
+  gboolean has_track_gain;
+  gboolean has_track_peak;
+  gboolean has_album_gain;
+  gboolean has_album_peak;
+};
+
+struct _GstRgAnalysisClass
+{
+  GstBaseTransformClass parent_class;
+};
+
+GType gst_rg_analysis_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_RG_ANALYSIS_H__ */
diff --git a/gst/replaygain/gstrglimiter.c b/gst/replaygain/gstrglimiter.c
new file mode 100644
index 0000000..95363be
--- /dev/null
+++ b/gst/replaygain/gstrglimiter.c
@@ -0,0 +1,201 @@
+/* GStreamer ReplayGain limiter
+ *
+ * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
+ * 
+ * gstrglimiter.c: Element to apply signal compression to raw audio data
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+/**
+ * SECTION:element-rglimiter
+ * @see_also: #GstRgVolume
+ *
+ * This element applies signal compression/limiting to raw audio data.  It
+ * performs strict hard limiting with soft-knee characteristics, using a
+ * threshold of -6 dB.  This type of filter is mentioned in the proposed <ulink
+ * url="http://replaygain.org">ReplayGain standard</ulink>.
+ * 
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch-1.0 filesrc location=filename.ext ! decodebin ! audioconvert \
+ *            ! rgvolume pre-amp=6.0 headroom=10.0 ! rglimiter \
+ *            ! audioconvert ! audioresample ! alsasink
+ * ]|Playback of a file
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+#include <math.h>
+#include <gst/audio/audio.h>
+
+#include "gstrglimiter.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_rg_limiter_debug);
+#define GST_CAT_DEFAULT gst_rg_limiter_debug
+
+enum
+{
+  PROP_0,
+  PROP_ENABLED,
+};
+
+static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " GST_AUDIO_NE (F32) ", "
+        "layout = (string) { interleaved, non-interleaved }, "
+        "channels = (int) [1, MAX], " "rate = (int) [1, MAX]"));
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " GST_AUDIO_NE (F32) ", "
+        "layout = (string) { interleaved, non-interleaved}, "
+        "channels = (int) [1, MAX], " "rate = (int) [1, MAX]"));
+
+#define gst_rg_limiter_parent_class parent_class
+G_DEFINE_TYPE (GstRgLimiter, gst_rg_limiter, GST_TYPE_BASE_TRANSFORM);
+
+static void gst_rg_limiter_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rg_limiter_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static GstFlowReturn gst_rg_limiter_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf);
+
+static void
+gst_rg_limiter_class_init (GstRgLimiterClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *element_class;
+  GstBaseTransformClass *trans_class;
+
+  gobject_class = (GObjectClass *) klass;
+  element_class = (GstElementClass *) klass;
+
+  gobject_class->set_property = gst_rg_limiter_set_property;
+  gobject_class->get_property = gst_rg_limiter_get_property;
+
+  g_object_class_install_property (gobject_class, PROP_ENABLED,
+      g_param_spec_boolean ("enabled", "Enabled", "Enable processing", TRUE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+
+  trans_class = GST_BASE_TRANSFORM_CLASS (klass);
+  trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_limiter_transform_ip);
+  trans_class->passthrough_on_same_caps = FALSE;
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_factory));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_factory));
+  gst_element_class_set_static_metadata (element_class, "ReplayGain limiter",
+      "Filter/Effect/Audio",
+      "Apply signal compression to raw audio data",
+      "Ren\xc3\xa9 Stadler <mail@renestadler.de>");
+
+  GST_DEBUG_CATEGORY_INIT (gst_rg_limiter_debug, "rglimiter", 0,
+      "ReplayGain limiter element");
+}
+
+static void
+gst_rg_limiter_init (GstRgLimiter * filter)
+{
+  GstBaseTransform *base = GST_BASE_TRANSFORM (filter);
+
+  gst_base_transform_set_passthrough (base, FALSE);
+  gst_base_transform_set_gap_aware (base, TRUE);
+
+  filter->enabled = TRUE;
+}
+
+static void
+gst_rg_limiter_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRgLimiter *filter = GST_RG_LIMITER (object);
+
+  switch (prop_id) {
+    case PROP_ENABLED:
+      filter->enabled = g_value_get_boolean (value);
+      gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
+          !filter->enabled);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rg_limiter_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRgLimiter *filter = GST_RG_LIMITER (object);
+
+  switch (prop_id) {
+    case PROP_ENABLED:
+      g_value_set_boolean (value, filter->enabled);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+#define LIMIT 1.0
+#define THRES 0.5               /* ca. -6 dB */
+#define COMPL 0.5               /* LIMIT - THRESH */
+
+static GstFlowReturn
+gst_rg_limiter_transform_ip (GstBaseTransform * base, GstBuffer * buf)
+{
+  GstRgLimiter *filter = GST_RG_LIMITER (base);
+  gfloat *input;
+  GstMapInfo map;
+  guint count;
+  guint i;
+
+  if (!filter->enabled)
+    return GST_FLOW_OK;
+
+  if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP))
+    return GST_FLOW_OK;
+
+  gst_buffer_map (buf, &map, GST_MAP_READ);
+  input = (gfloat *) map.data;
+  count = gst_buffer_get_size (buf) / sizeof (gfloat);
+
+  for (i = count; i--;) {
+    if (*input > THRES)
+      *input = tanhf ((*input - THRES) / COMPL) * COMPL + THRES;
+    else if (*input < -THRES)
+      *input = tanhf ((*input + THRES) / COMPL) * COMPL - THRES;
+    input++;
+  }
+
+  gst_buffer_unmap (buf, &map);
+
+  return GST_FLOW_OK;
+}
diff --git a/gst/replaygain/gstrglimiter.h b/gst/replaygain/gstrglimiter.h
new file mode 100644
index 0000000..63bd804
--- /dev/null
+++ b/gst/replaygain/gstrglimiter.h
@@ -0,0 +1,64 @@
+/* GStreamer ReplayGain limiter
+ *
+ * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
+ *
+ * gstrglimiter.h: Element to apply signal compression to raw audio data
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifndef __GST_RG_LIMITER_H__
+#define __GST_RG_LIMITER_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+
+#define GST_TYPE_RG_LIMITER \
+  (gst_rg_limiter_get_type())
+#define GST_RG_LIMITER(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_LIMITER,GstRgLimiter))
+#define GST_RG_LIMITER_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_LIMITER,GstRgLimiterClass))
+#define GST_IS_RG_LIMITER(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_LIMITER))
+#define GST_IS_RG_LIMITER_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_LIMITER))
+
+typedef struct _GstRgLimiter GstRgLimiter;
+typedef struct _GstRgLimiterClass GstRgLimiterClass;
+
+/**
+ * GstRgLimiter:
+ *
+ * Opaque data structure.
+ */
+struct _GstRgLimiter
+{
+  GstBaseTransform element;
+
+  /*< private >*/
+
+  gboolean enabled;
+};
+
+struct _GstRgLimiterClass
+{
+  GstBaseTransformClass parent_class;
+};
+
+GType gst_rg_limiter_get_type (void);
+
+#endif /* __GST_RG_LIMITER_H__ */
diff --git a/gst/replaygain/gstrgvolume.c b/gst/replaygain/gstrgvolume.c
new file mode 100755
index 0000000..beb8fda
--- /dev/null
+++ b/gst/replaygain/gstrgvolume.c
@@ -0,0 +1,689 @@
+/* GStreamer ReplayGain volume adjustment
+ *
+ * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
+ * 
+ * gstrgvolume.c: Element to apply ReplayGain volume adjustment
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+/**
+ * SECTION:element-rgvolume
+ * @see_also: #GstRgLimiter, #GstRgAnalysis
+ *
+ * This element applies volume changes to streams as lined out in the proposed
+ * <ulink url="http://replaygain.org">ReplayGain standard</ulink>.  It
+ * interprets the ReplayGain meta data tags and carries out the adjustment (by
+ * using a volume element internally).  The relevant tags are:
+ * <itemizedlist>
+ * <listitem>#GST_TAG_TRACK_GAIN</listitem>
+ * <listitem>#GST_TAG_TRACK_PEAK</listitem>
+ * <listitem>#GST_TAG_ALBUM_GAIN</listitem>
+ * <listitem>#GST_TAG_ALBUM_PEAK</listitem>
+ * <listitem>#GST_TAG_REFERENCE_LEVEL</listitem>
+ * </itemizedlist>
+ * The information carried by these tags must have been calculated beforehand by
+ * performing the ReplayGain analysis.  This is implemented by the <link
+ * linkend="GstRgAnalysis">rganalysis</link> element.
+ * 
+ * The signal compression/limiting recommendations outlined in the proposed
+ * standard are not implemented by this element.  This has to be handled by
+ * separate elements because applications might want to have additional filters
+ * between the volume adjustment and the limiting stage.  A basic limiter is
+ * included with this plugin: The <link linkend="GstRgLimiter">rglimiter</link>
+ * element applies -6 dB hard limiting as mentioned in the ReplayGain standard.
+ * 
+ * <refsect2>
+ * <title>Example launch line</title>
+ * |[
+ * gst-launch-1.0 filesrc location=filename.ext ! decodebin ! audioconvert \
+ *     ! rgvolume ! audioconvert ! audioresample ! alsasink
+ * ]| Playback of a file
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+#include <gst/pbutils/pbutils.h>
+#include <gst/audio/audio.h>
+#include <math.h>
+
+#include "gstrgvolume.h"
+#include "replaygain.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_rg_volume_debug);
+#define GST_CAT_DEFAULT gst_rg_volume_debug
+
+enum
+{
+  PROP_0,
+  PROP_ALBUM_MODE,
+  PROP_HEADROOM,
+  PROP_PRE_AMP,
+  PROP_FALLBACK_GAIN,
+  PROP_TARGET_GAIN,
+  PROP_RESULT_GAIN
+};
+
+#define DEFAULT_ALBUM_MODE TRUE
+#define DEFAULT_HEADROOM 0.0
+#define DEFAULT_PRE_AMP 0.0
+#define DEFAULT_FALLBACK_GAIN 0.0
+
+#define DB_TO_LINEAR(x) pow (10., (x) / 20.)
+#define LINEAR_TO_DB(x) (20. * log10 (x))
+
+#define GAIN_FORMAT "+.02f dB"
+#define PEAK_FORMAT ".06f"
+
+#define VALID_GAIN(x) ((x) > -60.00 && (x) < 60.00)
+#define VALID_PEAK(x) ((x) > 0.)
+
+/* Same template caps as GstVolume, for I don't like having just ANY caps. */
+
+#define FORMAT "{ "GST_AUDIO_NE(F32)","GST_AUDIO_NE(S16)" }"
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " FORMAT ", "
+        "layout = (string) { interleaved, non-interleaved }, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]"));
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS ("audio/x-raw, "
+        "format = (string) " FORMAT ", "
+        "layout = (string) { interleaved, non-interleaved }, "
+        "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]"));
+
+#define gst_rg_volume_parent_class parent_class
+G_DEFINE_TYPE (GstRgVolume, gst_rg_volume, GST_TYPE_BIN);
+
+static void gst_rg_volume_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rg_volume_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_rg_volume_dispose (GObject * object);
+
+static GstStateChangeReturn gst_rg_volume_change_state (GstElement * element,
+    GstStateChange transition);
+static gboolean gst_rg_volume_sink_event (GstPad * pad, GstObject * parent,
+    GstEvent * event);
+
+static GstEvent *gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event);
+static void gst_rg_volume_reset (GstRgVolume * self);
+static void gst_rg_volume_update_gain (GstRgVolume * self);
+static inline void gst_rg_volume_determine_gain (GstRgVolume * self,
+    gdouble * target_gain, gdouble * result_gain);
+
+static void
+gst_rg_volume_class_init (GstRgVolumeClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *element_class;
+  GstBinClass *bin_class;
+
+  gobject_class = (GObjectClass *) klass;
+
+  gobject_class->set_property = gst_rg_volume_set_property;
+  gobject_class->get_property = gst_rg_volume_get_property;
+  gobject_class->dispose = gst_rg_volume_dispose;
+
+  /**
+   * GstRgVolume:album-mode:
+   *
+   * Whether to prefer album gain over track gain.
+   *
+   * If set to %TRUE, use album gain instead of track gain if both are
+   * available.  This keeps the relative loudness levels of tracks from the same
+   * album intact.
+   *
+   * If set to %FALSE, track mode is used instead.  This effectively leads to
+   * more extensive normalization.
+   *
+   * If album mode is enabled but the album gain tag is absent in the stream,
+   * the track gain is used instead.  If both gain tags are missing, the value
+   * of the #GstRgVolume:fallback-gain property is used instead.
+   */
+  g_object_class_install_property (gobject_class, PROP_ALBUM_MODE,
+      g_param_spec_boolean ("album-mode", "Album mode",
+          "Prefer album over track gain", DEFAULT_ALBUM_MODE,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRgVolume:headroom:
+   *
+   * Extra headroom [dB].  This controls the amount by which the output can
+   * exceed digital full scale.
+   *
+   * Only set this to a value greater than 0.0 if signal compression/limiting of
+   * a suitable form is applied to the output (or output is brought into the
+   * correct range by some other transformation).
+   *
+   * This element internally uses a volume element, which also supports
+   * operating on integer audio formats.  These formats do not allow exceeding
+   * digital full scale.  If extra headroom is used, make sure that the raw
+   * audio data format is floating point (F32).  Otherwise,
+   * clipping distortion might be introduced as part of the volume adjustment
+   * itself.
+   */
+  g_object_class_install_property (gobject_class, PROP_HEADROOM,
+      g_param_spec_double ("headroom", "Headroom", "Extra headroom [dB]",
+          0., 60., DEFAULT_HEADROOM,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRgVolume:pre-amp:
+   *
+   * Additional gain to apply globally [dB].  This controls the trade-off
+   * between uniformity of normalization and utilization of available dynamic
+   * range.
+   *
+   * Note that the default value is 0 dB because the ReplayGain reference value
+   * was adjusted by +6 dB (from 83 to 89 dB).  At the time of this writing, the
+   * <ulink url="http://replaygain.org">webpage</ulink> is still outdated and
+   * does not reflect this change however.  Where the original proposal states
+   * that a proper default pre-amp value is +6 dB, this translates to the used 0
+   * dB.
+   */
+  g_object_class_install_property (gobject_class, PROP_PRE_AMP,
+      g_param_spec_double ("pre-amp", "Pre-amp", "Extra gain [dB]",
+          -60., 60., DEFAULT_PRE_AMP,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRgVolume:fallback-gain:
+   *
+   * Fallback gain [dB] for streams missing ReplayGain tags.
+   */
+  g_object_class_install_property (gobject_class, PROP_FALLBACK_GAIN,
+      g_param_spec_double ("fallback-gain", "Fallback gain",
+          "Gain for streams missing tags [dB]",
+          -60., 60., DEFAULT_FALLBACK_GAIN,
+          G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRgVolume:result-gain:
+   *
+   * Applied gain [dB].  This gain is applied to processed buffer data.
+   *
+   * This is set to the #GstRgVolume:target-gain if amplification by that amount
+   * can be applied safely. "Safely" means that the volume adjustment does not
+   * inflict clipping distortion.  Should this not be the case, the result gain
+   * is set to an appropriately reduced value (by applying peak normalization).
+   * The proposed standard calls this "clipping prevention".
+   *
+   * The difference between target and result gain reflects the necessary amount
+   * of reduction.  Applications can make use of this information to temporarily
+   * reduce the #GstRgVolume:pre-amp for subsequent streams, as recommended by
+   * the ReplayGain standard.
+   *
+   * Note that target and result gain differing for a great majority of streams
+   * indicates a problem: What happens in this case is that most streams receive
+   * peak normalization instead of amplification by the ideal replay gain.  To
+   * prevent this, the #GstRgVolume:pre-amp has to be lowered and/or a limiter
+   * has to be used which facilitates the use of #GstRgVolume:headroom.
+   */
+  g_object_class_install_property (gobject_class, PROP_RESULT_GAIN,
+      g_param_spec_double ("result-gain", "Result-gain", "Applied gain [dB]",
+          -120., 120., 0., G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+  /**
+   * GstRgVolume:target-gain:
+   *
+   * Applicable gain [dB].  This gain is supposed to be applied.
+   *
+   * Depending on the value of the #GstRgVolume:album-mode property and the
+   * presence of ReplayGain tags in the stream, this is set according to one of
+   * these simple formulas:
+   *
+   * <itemizedlist>
+   * <listitem>#GstRgVolume:pre-amp + album gain of the stream</listitem>
+   * <listitem>#GstRgVolume:pre-amp + track gain of the stream</listitem>
+   * <listitem>#GstRgVolume:pre-amp + #GstRgVolume:fallback-gain</listitem>
+   * </itemizedlist>
+   */
+  g_object_class_install_property (gobject_class, PROP_TARGET_GAIN,
+      g_param_spec_double ("target-gain", "Target-gain",
+          "Applicable gain [dB]", -120., 120., 0.,
+          G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
+
+  element_class = (GstElementClass *) klass;
+  element_class->change_state = GST_DEBUG_FUNCPTR (gst_rg_volume_change_state);
+
+  bin_class = (GstBinClass *) klass;
+  /* Setting these to NULL makes gst_bin_add and _remove refuse to let anyone
+   * mess with our internals. */
+  bin_class->add_element = NULL;
+  bin_class->remove_element = NULL;
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_template));
+  gst_element_class_set_static_metadata (element_class, "ReplayGain volume",
+      "Filter/Effect/Audio",
+      "Apply ReplayGain volume adjustment",
+      "Ren\xc3\xa9 Stadler <mail@renestadler.de>");
+
+  GST_DEBUG_CATEGORY_INIT (gst_rg_volume_debug, "rgvolume", 0,
+      "ReplayGain volume element");
+}
+
+static void
+gst_rg_volume_init (GstRgVolume * self)
+{
+  GObjectClass *volume_class;
+  GstPad *volume_pad, *ghost_pad;
+
+  self->album_mode = DEFAULT_ALBUM_MODE;
+  self->headroom = DEFAULT_HEADROOM;
+  self->pre_amp = DEFAULT_PRE_AMP;
+  self->fallback_gain = DEFAULT_FALLBACK_GAIN;
+  self->target_gain = 0.0;
+  self->result_gain = 0.0;
+
+  self->volume_element = gst_element_factory_make ("volume", "rgvolume-volume");
+  if (G_UNLIKELY (self->volume_element == NULL)) {
+    GstMessage *msg;
+
+    GST_WARNING_OBJECT (self, "could not create volume element");
+    msg = gst_missing_element_message_new (GST_ELEMENT_CAST (self), "volume");
+    gst_element_post_message (GST_ELEMENT_CAST (self), msg);
+
+    /* Nothing else to do, we will refuse the state change from NULL to READY to
+     * indicate that something went very wrong.  It is doubtful that someone
+     * attempts changing our state though, since we end up having no pads! */
+    return;
+  }
+
+  volume_class = G_OBJECT_GET_CLASS (G_OBJECT (self->volume_element));
+  self->max_volume = G_PARAM_SPEC_DOUBLE
+      (g_object_class_find_property (volume_class, "volume"))->maximum;
+
+  GST_BIN_CLASS (parent_class)->add_element (GST_BIN_CAST (self),
+      self->volume_element);
+
+  volume_pad = gst_element_get_static_pad (self->volume_element, "sink");
+  ghost_pad = gst_ghost_pad_new_from_template ("sink", volume_pad,
+      GST_PAD_PAD_TEMPLATE (volume_pad));
+  gst_object_unref (volume_pad);
+  gst_pad_set_event_function (ghost_pad, gst_rg_volume_sink_event);
+  gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad);
+
+  volume_pad = gst_element_get_static_pad (self->volume_element, "src");
+  ghost_pad = gst_ghost_pad_new_from_template ("src", volume_pad,
+      GST_PAD_PAD_TEMPLATE (volume_pad));
+  gst_object_unref (volume_pad);
+  gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad);
+}
+
+static void
+gst_rg_volume_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRgVolume *self = GST_RG_VOLUME (object);
+
+  switch (prop_id) {
+    case PROP_ALBUM_MODE:
+      self->album_mode = g_value_get_boolean (value);
+      break;
+    case PROP_HEADROOM:
+      self->headroom = g_value_get_double (value);
+      break;
+    case PROP_PRE_AMP:
+      self->pre_amp = g_value_get_double (value);
+      break;
+    case PROP_FALLBACK_GAIN:
+      self->fallback_gain = g_value_get_double (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+
+  gst_rg_volume_update_gain (self);
+}
+
+static void
+gst_rg_volume_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRgVolume *self = GST_RG_VOLUME (object);
+
+  switch (prop_id) {
+    case PROP_ALBUM_MODE:
+      g_value_set_boolean (value, self->album_mode);
+      break;
+    case PROP_HEADROOM:
+      g_value_set_double (value, self->headroom);
+      break;
+    case PROP_PRE_AMP:
+      g_value_set_double (value, self->pre_amp);
+      break;
+    case PROP_FALLBACK_GAIN:
+      g_value_set_double (value, self->fallback_gain);
+      break;
+    case PROP_TARGET_GAIN:
+      g_value_set_double (value, self->target_gain);
+      break;
+    case PROP_RESULT_GAIN:
+      g_value_set_double (value, self->result_gain);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rg_volume_dispose (GObject * object)
+{
+  GstRgVolume *self = GST_RG_VOLUME (object);
+
+  if (self->volume_element != NULL) {
+    /* Manually remove our child using the bin implementation of remove_element.
+     * This is needed because we prevent gst_bin_remove from working, which the
+     * parent dispose handler would use if we had any children left. */
+    GST_BIN_CLASS (parent_class)->remove_element (GST_BIN_CAST (self),
+        self->volume_element);
+    self->volume_element = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static GstStateChangeReturn
+gst_rg_volume_change_state (GstElement * element, GstStateChange transition)
+{
+  GstRgVolume *self = GST_RG_VOLUME (element);
+  GstStateChangeReturn res;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+
+      if (G_UNLIKELY (self->volume_element == NULL)) {
+        /* Creating our child volume element in _init failed. */
+        return GST_STATE_CHANGE_FAILURE;
+      }
+      break;
+
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+
+      gst_rg_volume_reset (self);
+      break;
+
+    default:
+      break;
+  }
+
+  res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  return res;
+}
+
+/* Event function for the ghost sink pad. */
+static gboolean
+gst_rg_volume_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
+{
+  GstRgVolume *self;
+  GstEvent *send_event = event;
+  gboolean res;
+
+  self = GST_RG_VOLUME (parent);
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_TAG:
+
+      GST_LOG_OBJECT (self, "received tag event");
+
+      send_event = gst_rg_volume_tag_event (self, event);
+
+      if (send_event == NULL)
+        GST_LOG_OBJECT (self, "all tags handled, dropping event");
+
+      break;
+
+    case GST_EVENT_EOS:
+
+      gst_rg_volume_reset (self);
+      break;
+
+    default:
+      break;
+  }
+
+  if (G_LIKELY (send_event != NULL))
+    res = gst_pad_event_default (pad, parent, send_event);
+  else
+    res = TRUE;
+
+  return res;
+}
+
+static GstEvent *
+gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event)
+{
+  GstTagList *tag_list;
+  gboolean has_track_gain, has_track_peak, has_album_gain, has_album_peak;
+  gboolean has_ref_level;
+
+  g_return_val_if_fail (event != NULL, NULL);
+  g_return_val_if_fail (GST_EVENT_TYPE (event) == GST_EVENT_TAG, event);
+
+  gst_event_parse_tag (event, &tag_list);
+
+  if (gst_tag_list_is_empty (tag_list))
+    return event;
+
+  has_track_gain = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN,
+      &self->track_gain);
+  has_track_peak = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK,
+      &self->track_peak);
+  has_album_gain = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN,
+      &self->album_gain);
+  has_album_peak = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK,
+      &self->album_peak);
+  has_ref_level = gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
+      &self->reference_level);
+
+  if (!has_track_gain && !has_track_peak && !has_album_gain && !has_album_peak)
+    return event;
+
+  if (has_ref_level && (has_track_gain || has_album_gain)
+      && (ABS (self->reference_level - RG_REFERENCE_LEVEL) > 1.e-6)) {
+    /* Log a message stating the amount of adjustment that is applied below. */
+    GST_DEBUG_OBJECT (self,
+        "compensating for reference level difference by %" GAIN_FORMAT,
+        RG_REFERENCE_LEVEL - self->reference_level);
+  }
+  if (has_track_gain) {
+    self->track_gain += RG_REFERENCE_LEVEL - self->reference_level;
+  }
+  if (has_album_gain) {
+    self->album_gain += RG_REFERENCE_LEVEL - self->reference_level;
+  }
+
+  /* Ignore values that are obviously invalid. */
+  if (G_UNLIKELY (has_track_gain && !VALID_GAIN (self->track_gain))) {
+    GST_DEBUG_OBJECT (self,
+        "ignoring bogus track gain value %" GAIN_FORMAT, self->track_gain);
+    has_track_gain = FALSE;
+  }
+  if (G_UNLIKELY (has_track_peak && !VALID_PEAK (self->track_peak))) {
+    GST_DEBUG_OBJECT (self,
+        "ignoring bogus track peak value %" PEAK_FORMAT, self->track_peak);
+    has_track_peak = FALSE;
+  }
+  if (G_UNLIKELY (has_album_gain && !VALID_GAIN (self->album_gain))) {
+    GST_DEBUG_OBJECT (self,
+        "ignoring bogus album gain value %" GAIN_FORMAT, self->album_gain);
+    has_album_gain = FALSE;
+  }
+  if (G_UNLIKELY (has_album_peak && !VALID_PEAK (self->album_peak))) {
+    GST_DEBUG_OBJECT (self,
+        "ignoring bogus album peak value %" PEAK_FORMAT, self->album_peak);
+    has_album_peak = FALSE;
+  }
+
+  /* Clamp peaks >1.0.  Float based decoders can produce spurious samples >1.0,
+   * cutting these files back to 1.0 should not cause any audible distortion.
+   * This is most often seen with Vorbis files. */
+  if (has_track_peak && self->track_peak > 1.) {
+    GST_DEBUG_OBJECT (self,
+        "clamping track peak %" PEAK_FORMAT " to 1.0", self->track_peak);
+    self->track_peak = 1.0;
+  }
+  if (has_album_peak && self->album_peak > 1.) {
+    GST_DEBUG_OBJECT (self,
+        "clamping album peak %" PEAK_FORMAT " to 1.0", self->album_peak);
+    self->album_peak = 1.0;
+  }
+
+  self->has_track_gain |= has_track_gain;
+  self->has_track_peak |= has_track_peak;
+  self->has_album_gain |= has_album_gain;
+  self->has_album_peak |= has_album_peak;
+
+  event = (GstEvent *) gst_mini_object_make_writable (GST_MINI_OBJECT (event));
+  gst_event_parse_tag (event, &tag_list);
+
+  gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_GAIN);
+  gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_PEAK);
+  gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_GAIN);
+  gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_PEAK);
+  gst_tag_list_remove_tag (tag_list, GST_TAG_REFERENCE_LEVEL);
+
+  gst_rg_volume_update_gain (self);
+
+  if (gst_tag_list_is_empty (tag_list)) {
+    gst_event_unref (event);
+    event = NULL;
+  }
+
+  return event;
+}
+
+static void
+gst_rg_volume_reset (GstRgVolume * self)
+{
+  self->has_track_gain = FALSE;
+  self->has_track_peak = FALSE;
+  self->has_album_gain = FALSE;
+  self->has_album_peak = FALSE;
+
+  self->reference_level = RG_REFERENCE_LEVEL;
+
+  gst_rg_volume_update_gain (self);
+}
+
+static void
+gst_rg_volume_update_gain (GstRgVolume * self)
+{
+  gdouble target_gain, result_gain, result_volume;
+  gboolean target_gain_changed, result_gain_changed;
+
+  gst_rg_volume_determine_gain (self, &target_gain, &result_gain);
+
+  result_volume = DB_TO_LINEAR (result_gain);
+
+  /* Ensure that the result volume is within the range that the volume element
+   * can handle.  Currently, the limit is 10. (+20 dB), which should not be
+   * restrictive. */
+  if (G_UNLIKELY (result_volume > self->max_volume)) {
+    GST_INFO_OBJECT (self,
+        "cannot handle result gain of %" GAIN_FORMAT " (%0.6f), adjusting",
+        result_gain, result_volume);
+
+    result_volume = self->max_volume;
+    result_gain = LINEAR_TO_DB (result_volume);
+  }
+
+  /* Direct comparison is OK in this case. */
+  if (target_gain == result_gain) {
+    GST_INFO_OBJECT (self,
+        "result gain is %" GAIN_FORMAT " (%0.6f), matching target",
+        result_gain, result_volume);
+  } else {
+    GST_INFO_OBJECT (self,
+        "result gain is %" GAIN_FORMAT " (%0.6f), target is %" GAIN_FORMAT,
+        result_gain, result_volume, target_gain);
+  }
+
+  target_gain_changed = (self->target_gain != target_gain);
+  result_gain_changed = (self->result_gain != result_gain);
+
+  self->target_gain = target_gain;
+  self->result_gain = result_gain;
+
+  g_object_set (self->volume_element, "volume", result_volume, NULL);
+
+  if (target_gain_changed)
+    g_object_notify ((GObject *) self, "target-gain");
+  if (result_gain_changed)
+    g_object_notify ((GObject *) self, "result-gain");
+}
+
+static inline void
+gst_rg_volume_determine_gain (GstRgVolume * self, gdouble * target_gain,
+    gdouble * result_gain)
+{
+  gdouble gain, peak;
+
+  if (!self->has_track_gain && !self->has_album_gain) {
+
+    GST_DEBUG_OBJECT (self, "using fallback gain");
+    gain = self->fallback_gain;
+    peak = 1.0;
+
+  } else if ((self->album_mode && self->has_album_gain)
+      || (!self->album_mode && !self->has_track_gain)) {
+
+    gain = self->album_gain;
+    if (G_LIKELY (self->has_album_peak)) {
+      peak = self->album_peak;
+    } else {
+      GST_DEBUG_OBJECT (self, "album peak missing, assuming 1.0");
+      peak = 1.0;
+    }
+    /* Falling back from track to album gain shouldn't really happen. */
+    if (G_UNLIKELY (!self->album_mode))
+      GST_INFO_OBJECT (self, "falling back to album gain");
+
+  } else {
+    /* !album_mode && !has_album_gain || album_mode && has_track_gain */
+
+    gain = self->track_gain;
+    if (G_LIKELY (self->has_track_peak)) {
+      peak = self->track_peak;
+    } else {
+      GST_DEBUG_OBJECT (self, "track peak missing, assuming 1.0");
+      peak = 1.0;
+    }
+    if (self->album_mode)
+      GST_INFO_OBJECT (self, "falling back to track gain");
+  }
+
+  gain += self->pre_amp;
+
+  *target_gain = gain;
+  *result_gain = gain;
+
+  if (LINEAR_TO_DB (peak) + gain > self->headroom) {
+    *result_gain = LINEAR_TO_DB (1. / peak) + self->headroom;
+  }
+}
diff --git a/gst/replaygain/gstrgvolume.h b/gst/replaygain/gstrgvolume.h
new file mode 100644
index 0000000..a0a5a8c
--- /dev/null
+++ b/gst/replaygain/gstrgvolume.h
@@ -0,0 +1,88 @@
+/* GStreamer ReplayGain volume adjustment
+ *
+ * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
+ *
+ * gstrgvolume.h: Element to apply ReplayGain volume adjustment
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifndef __GST_RG_VOLUME_H__
+#define __GST_RG_VOLUME_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RG_VOLUME \
+  (gst_rg_volume_get_type())
+#define GST_RG_VOLUME(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_VOLUME,GstRgVolume))
+#define GST_RG_VOLUME_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_VOLUME,GstRgVolumeClass))
+#define GST_IS_RG_VOLUME(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_VOLUME))
+#define GST_IS_RG_VOLUME_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_VOLUME))
+
+typedef struct _GstRgVolume GstRgVolume;
+typedef struct _GstRgVolumeClass GstRgVolumeClass;
+
+/**
+ * GstRgVolume:
+ *
+ * Opaque data structure.
+ */
+struct _GstRgVolume
+{
+  GstBin bin;
+
+  /*< private >*/
+
+  GstElement *volume_element;
+  gdouble max_volume;
+
+  gboolean album_mode;
+  gdouble headroom;
+  gdouble pre_amp;
+  gdouble fallback_gain;
+
+  gdouble target_gain;
+  gdouble result_gain;
+
+  gdouble track_gain;
+  gdouble track_peak;
+  gdouble album_gain;
+  gdouble album_peak;
+
+  gboolean has_track_gain;
+  gboolean has_track_peak;
+  gboolean has_album_gain;
+  gboolean has_album_peak;
+
+  gdouble reference_level;
+};
+
+struct _GstRgVolumeClass
+{
+  GstBinClass parent_class;
+};
+
+GType gst_rg_volume_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_RG_VOLUME_H__ */
diff --git a/gst/replaygain/replaygain.c b/gst/replaygain/replaygain.c
new file mode 100644
index 0000000..cb499ed
--- /dev/null
+++ b/gst/replaygain/replaygain.c
@@ -0,0 +1,53 @@
+/* GStreamer ReplayGain plugin
+ *
+ * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
+ * 
+ * replaygain.c: Plugin providing ReplayGain related elements
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+
+#include "gstrganalysis.h"
+#include "gstrglimiter.h"
+#include "gstrgvolume.h"
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  if (!gst_element_register (plugin, "rganalysis", GST_RANK_NONE,
+          GST_TYPE_RG_ANALYSIS))
+    return FALSE;
+
+  if (!gst_element_register (plugin, "rglimiter", GST_RANK_NONE,
+          GST_TYPE_RG_LIMITER))
+    return FALSE;
+
+  if (!gst_element_register (plugin, "rgvolume", GST_RANK_NONE,
+          GST_TYPE_RG_VOLUME))
+    return FALSE;
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, replaygain,
+    "ReplayGain volume normalization", plugin_init, VERSION, GST_LICENSE,
+    GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
diff --git a/gst/replaygain/replaygain.h b/gst/replaygain/replaygain.h
new file mode 100644
index 0000000..15be888
--- /dev/null
+++ b/gst/replaygain/replaygain.h
@@ -0,0 +1,36 @@
+/* GStreamer ReplayGain plugin
+ *
+ * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
+ *
+ * replaygain.h: Plugin providing ReplayGain related elements
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifndef __REPLAYGAIN_H__
+#define __REPLAYGAIN_H__
+
+G_BEGIN_DECLS
+
+/* Reference level (in dBSPL).  The 2001 proposal specifies 83.  This was
+ * changed later in all implementations to 89, which is the new, offical value:
+ * David Robinson acknowledged the change but didn't update the website yet. */
+
+#define RG_REFERENCE_LEVEL 89.
+
+G_END_DECLS
+
+#endif /* __REPLAYGAIN_H__ */
diff --git a/gst/replaygain/rganalysis.c b/gst/replaygain/rganalysis.c
new file mode 100644
index 0000000..2553536
--- /dev/null
+++ b/gst/replaygain/rganalysis.c
@@ -0,0 +1,824 @@
+/* GStreamer ReplayGain analysis
+ *
+ * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
+ * Copyright (C) 2001 David Robinson <David@Robinson.org>
+ *                    Glen Sawyer <glensawyer@hotmail.com>
+ *
+ * rganalysis.c: Analyze raw audio data in accordance with ReplayGain
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+/* Based on code with Copyright (C) 2001 David Robinson
+ * <David@Robinson.org> and Glen Sawyer <glensawyer@hotmail.com>,
+ * which is distributed under the LGPL as part of the vorbisgain
+ * program.  The original code also mentions Frank Klemm
+ * (http://www.uni-jena.de/~pfk/mpp/) for having contributed lots of
+ * good code.  Specifically, this is based on the file
+ * "gain_analysis.c" from vorbisgain version 0.34.
+ */
+
+/* Room for future improvement: Mono data is currently in fact copied
+ * to two channels which get processed normally.  This means that mono
+ * input data is processed twice.
+ */
+
+/* Helpful information for understanding this code: The two IIR
+ * filters depend on previous input _and_ previous output samples (up
+ * to the filter's order number of samples).  This explains the whole
+ * lot of memcpy'ing done in rg_analysis_analyze and why the context
+ * holds so many buffers.
+ */
+
+#include <math.h>
+#include <string.h>
+#include <glib.h>
+
+#include "rganalysis.h"
+
+#define YULE_ORDER         10
+#define BUTTER_ORDER        2
+/* Percentile which is louder than the proposed level: */
+#define RMS_PERCENTILE     95
+/* Duration of RMS window in milliseconds: */
+#define RMS_WINDOW_MSECS   50
+/* Histogram array elements per dB: */
+#define STEPS_PER_DB      100
+/* Histogram upper bound in dB (normal max. values in the wild are
+ * assumed to be around 70, 80 dB): */
+#define MAX_DB            120
+/* Calibration value: */
+#define PINK_REF           64.82        /* 298640883795 */
+
+#define MAX_ORDER         MAX (BUTTER_ORDER, YULE_ORDER)
+#define MAX_SAMPLE_RATE   48000
+/* The + 999 has the effect of ceil()ing: */
+#define MAX_SAMPLE_WINDOW (guint) \
+  ((MAX_SAMPLE_RATE * RMS_WINDOW_MSECS + 999) / 1000)
+
+/* Analysis result accumulator. */
+
+struct _RgAnalysisAcc
+{
+  guint32 histogram[STEPS_PER_DB * MAX_DB];
+  gdouble peak;
+};
+
+typedef struct _RgAnalysisAcc RgAnalysisAcc;
+
+/* Analysis context. */
+
+struct _RgAnalysisCtx
+{
+  /* Filter buffers for left channel. */
+  gfloat inprebuf_l[MAX_ORDER * 2];
+  gfloat *inpre_l;
+  gfloat stepbuf_l[MAX_SAMPLE_WINDOW + MAX_ORDER];
+  gfloat *step_l;
+  gfloat outbuf_l[MAX_SAMPLE_WINDOW + MAX_ORDER];
+  gfloat *out_l;
+  /* Filter buffers for right channel. */
+  gfloat inprebuf_r[MAX_ORDER * 2];
+  gfloat *inpre_r;
+  gfloat stepbuf_r[MAX_SAMPLE_WINDOW + MAX_ORDER];
+  gfloat *step_r;
+  gfloat outbuf_r[MAX_SAMPLE_WINDOW + MAX_ORDER];
+  gfloat *out_r;
+
+  /* Number of samples to reach duration of the RMS window: */
+  guint window_n_samples;
+  /* Progress of the running window: */
+  guint window_n_samples_done;
+  gdouble window_square_sum;
+
+  gint sample_rate;
+  gint sample_rate_index;
+
+  RgAnalysisAcc track;
+  RgAnalysisAcc album;
+  void (*post_message) (gpointer analysis,
+      GstClockTime timestamp, GstClockTime duration, gdouble rglevel);
+  gpointer analysis;
+  /* The timestamp of the current incoming buffer. */
+  GstClockTime buffer_timestamp;
+  /* Number of samples processed in current buffer, during emit_signal,
+     this will always be on an RMS window boundary. */
+  guint buffer_n_samples_done;
+};
+
+/* Filter coefficients for the IIR filters that form the equal
+ * loudness filter.  XFilter[ctx->sample_rate_index] gives the array
+ * of the X coefficients (A or B) for the configured sample rate. */
+
+#ifdef _MSC_VER
+/* Disable double-to-float warning: */
+/* A better solution would be to append 'f' to each constant, but that
+ * makes the code ugly. */
+#pragma warning ( disable : 4305 )
+#endif
+
+static const gfloat AYule[9][11] = {
+  {1., -3.84664617118067, 7.81501653005538, -11.34170355132042,
+        13.05504219327545, -12.28759895145294, 9.48293806319790,
+        -5.87257861775999, 2.75465861874613, -0.86984376593551,
+      0.13919314567432},
+  {1., -3.47845948550071, 6.36317777566148, -8.54751527471874, 9.47693607801280,
+        -8.81498681370155, 6.85401540936998, -4.39470996079559,
+      2.19611684890774, -0.75104302451432, 0.13149317958808},
+  {1., -2.37898834973084, 2.84868151156327, -2.64577170229825, 2.23697657451713,
+        -1.67148153367602, 1.00595954808547, -0.45953458054983,
+      0.16378164858596, -0.05032077717131, 0.02347897407020},
+  {1., -1.61273165137247, 1.07977492259970, -0.25656257754070,
+        -0.16276719120440, -0.22638893773906, 0.39120800788284,
+        -0.22138138954925, 0.04500235387352, 0.02005851806501,
+      0.00302439095741},
+  {1., -1.49858979367799, 0.87350271418188, 0.12205022308084, -0.80774944671438,
+        0.47854794562326, -0.12453458140019, -0.04067510197014,
+      0.08333755284107, -0.04237348025746, 0.02977207319925},
+  {1., -0.62820619233671, 0.29661783706366, -0.37256372942400, 0.00213767857124,
+        -0.42029820170918, 0.22199650564824, 0.00613424350682, 0.06747620744683,
+      0.05784820375801, 0.03222754072173},
+  {1., -1.04800335126349, 0.29156311971249, -0.26806001042947, 0.00819999645858,
+        0.45054734505008, -0.33032403314006, 0.06739368333110,
+      -0.04784254229033, 0.01639907836189, 0.01807364323573},
+  {1., -0.51035327095184, -0.31863563325245, -0.20256413484477,
+        0.14728154134330, 0.38952639978999, -0.23313271880868,
+        -0.05246019024463, -0.02505961724053, 0.02442357316099,
+      0.01818801111503},
+  {1., -0.25049871956020, -0.43193942311114, -0.03424681017675,
+        -0.04678328784242, 0.26408300200955, 0.15113130533216,
+        -0.17556493366449, -0.18823009262115, 0.05477720428674,
+      0.04704409688120}
+};
+
+static const gfloat BYule[9][11] = {
+  {0.03857599435200, -0.02160367184185, -0.00123395316851, -0.00009291677959,
+        -0.01655260341619, 0.02161526843274, -0.02074045215285,
+      0.00594298065125, 0.00306428023191, 0.00012025322027, 0.00288463683916},
+  {0.05418656406430, -0.02911007808948, -0.00848709379851, -0.00851165645469,
+        -0.00834990904936, 0.02245293253339, -0.02596338512915,
+        0.01624864962975, -0.00240879051584, 0.00674613682247,
+      -0.00187763777362},
+  {0.15457299681924, -0.09331049056315, -0.06247880153653, 0.02163541888798,
+        -0.05588393329856, 0.04781476674921, 0.00222312597743, 0.03174092540049,
+      -0.01390589421898, 0.00651420667831, -0.00881362733839},
+  {0.30296907319327, -0.22613988682123, -0.08587323730772, 0.03282930172664,
+        -0.00915702933434, -0.02364141202522, -0.00584456039913,
+        0.06276101321749, -0.00000828086748, 0.00205861885564,
+      -0.02950134983287},
+  {0.33642304856132, -0.25572241425570, -0.11828570177555, 0.11921148675203,
+        -0.07834489609479, -0.00469977914380, -0.00589500224440,
+        0.05724228140351, 0.00832043980773, -0.01635381384540,
+      -0.01760176568150},
+  {0.44915256608450, -0.14351757464547, -0.22784394429749, -0.01419140100551,
+        0.04078262797139, -0.12398163381748, 0.04097565135648, 0.10478503600251,
+      -0.01863887810927, -0.03193428438915, 0.00541907748707},
+  {0.56619470757641, -0.75464456939302, 0.16242137742230, 0.16744243493672,
+        -0.18901604199609, 0.30931782841830, -0.27562961986224,
+        0.00647310677246, 0.08647503780351, -0.03788984554840,
+      -0.00588215443421},
+  {0.58100494960553, -0.53174909058578, -0.14289799034253, 0.17520704835522,
+        0.02377945217615, 0.15558449135573, -0.25344790059353, 0.01628462406333,
+      0.06920467763959, -0.03721611395801, -0.00749618797172},
+  {0.53648789255105, -0.42163034350696, -0.00275953611929, 0.04267842219415,
+        -0.10214864179676, 0.14590772289388, -0.02459864859345,
+        -0.11202315195388, -0.04060034127000, 0.04788665548180,
+      -0.02217936801134}
+};
+
+static const gfloat AButter[9][3] = {
+  {1., -1.97223372919527, 0.97261396931306},
+  {1., -1.96977855582618, 0.97022847566350},
+  {1., -1.95835380975398, 0.95920349965459},
+  {1., -1.95002759149878, 0.95124613669835},
+  {1., -1.94561023566527, 0.94705070426118},
+  {1., -1.92783286977036, 0.93034775234268},
+  {1., -1.91858953033784, 0.92177618768381},
+  {1., -1.91542108074780, 0.91885558323625},
+  {1., -1.88903307939452, 0.89487434461664}
+};
+
+static const gfloat BButter[9][3] = {
+  {0.98621192462708, -1.97242384925416, 0.98621192462708},
+  {0.98500175787242, -1.97000351574484, 0.98500175787242},
+  {0.97938932735214, -1.95877865470428, 0.97938932735214},
+  {0.97531843204928, -1.95063686409857, 0.97531843204928},
+  {0.97316523498161, -1.94633046996323, 0.97316523498161},
+  {0.96454515552826, -1.92909031105652, 0.96454515552826},
+  {0.96009142950541, -1.92018285901082, 0.96009142950541},
+  {0.95856916599601, -1.91713833199203, 0.95856916599601},
+  {0.94597685600279, -1.89195371200558, 0.94597685600279}
+};
+
+#ifdef _MSC_VER
+#pragma warning ( default : 4305 )
+#endif
+
+/* Filter functions.  These access elements with negative indices of
+ * the input and output arrays (up to the filter's order). */
+
+/* For much better performance, the function below has been
+ * implemented by unrolling the inner loop for our two use cases. */
+
+/*
+ * static inline void
+ * apply_filter (const gfloat * input, gfloat * output, guint n_samples,
+ *     const gfloat * a, const gfloat * b, guint order)
+ * {
+ *   gfloat y;
+ *   gint i, k;
+ * 
+ *   for (i = 0; i < n_samples; i++) {
+ *     y = input[i] * b[0];
+ *     for (k = 1; k <= order; k++)
+ *       y += input[i - k] * b[k] - output[i - k] * a[k];
+ *     output[i] = y;
+ *   }
+ * }
+ */
+
+static inline void
+yule_filter (const gfloat * input, gfloat * output,
+    const gfloat * a, const gfloat * b)
+{
+  /* 1e-10 is added below to avoid running into denormals when operating on
+   * near silence. */
+
+  output[0] = 1e-10 + input[0] * b[0]
+      + input[-1] * b[1] - output[-1] * a[1]
+      + input[-2] * b[2] - output[-2] * a[2]
+      + input[-3] * b[3] - output[-3] * a[3]
+      + input[-4] * b[4] - output[-4] * a[4]
+      + input[-5] * b[5] - output[-5] * a[5]
+      + input[-6] * b[6] - output[-6] * a[6]
+      + input[-7] * b[7] - output[-7] * a[7]
+      + input[-8] * b[8] - output[-8] * a[8]
+      + input[-9] * b[9] - output[-9] * a[9]
+      + input[-10] * b[10] - output[-10] * a[10];
+}
+
+static inline void
+butter_filter (const gfloat * input, gfloat * output,
+    const gfloat * a, const gfloat * b)
+{
+  output[0] = input[0] * b[0]
+      + input[-1] * b[1] - output[-1] * a[1]
+      + input[-2] * b[2] - output[-2] * a[2];
+}
+
+/* Because butter_filter and yule_filter are inlined, this function is
+ * a bit blown-up (code-size wise), but not inlining gives a ca. 40%
+ * performance penalty. */
+
+static inline void
+apply_filters (const RgAnalysisCtx * ctx, const gfloat * input_l,
+    const gfloat * input_r, guint n_samples)
+{
+  const gfloat *ayule = AYule[ctx->sample_rate_index];
+  const gfloat *byule = BYule[ctx->sample_rate_index];
+  const gfloat *abutter = AButter[ctx->sample_rate_index];
+  const gfloat *bbutter = BButter[ctx->sample_rate_index];
+  gint pos = ctx->window_n_samples_done;
+  gint i;
+
+  for (i = 0; i < n_samples; i++, pos++) {
+    yule_filter (input_l + i, ctx->step_l + pos, ayule, byule);
+    butter_filter (ctx->step_l + pos, ctx->out_l + pos, abutter, bbutter);
+
+    yule_filter (input_r + i, ctx->step_r + pos, ayule, byule);
+    butter_filter (ctx->step_r + pos, ctx->out_r + pos, abutter, bbutter);
+  }
+}
+
+/* Clear filter buffer state and current RMS window. */
+
+static void
+reset_filters (RgAnalysisCtx * ctx)
+{
+  gint i;
+
+  for (i = 0; i < MAX_ORDER; i++) {
+
+    ctx->inprebuf_l[i] = 0.;
+    ctx->stepbuf_l[i] = 0.;
+    ctx->outbuf_l[i] = 0.;
+
+    ctx->inprebuf_r[i] = 0.;
+    ctx->stepbuf_r[i] = 0.;
+    ctx->outbuf_r[i] = 0.;
+  }
+
+  ctx->window_square_sum = 0.;
+  ctx->window_n_samples_done = 0;
+}
+
+/* Accumulator functions. */
+
+/* Add two accumulators in-place.  The sum is defined as the result of
+ * the vector sum of the histogram array and the maximum value of the
+ * peak field.  Thus "adding" the accumulators for all tracks yields
+ * the correct result for obtaining the album gain and peak. */
+
+static void
+accumulator_add (RgAnalysisAcc * acc, const RgAnalysisAcc * acc_other)
+{
+  gint i;
+
+  for (i = 0; i < G_N_ELEMENTS (acc->histogram); i++)
+    acc->histogram[i] += acc_other->histogram[i];
+
+  acc->peak = MAX (acc->peak, acc_other->peak);
+}
+
+/* Reset an accumulator to zero. */
+
+static void
+accumulator_clear (RgAnalysisAcc * acc)
+{
+  memset (acc->histogram, 0, sizeof (acc->histogram));
+  acc->peak = 0.;
+}
+
+/* Obtain final analysis result from an accumulator.  Returns TRUE on
+ * success, FALSE on error (if accumulator is still zero). */
+
+static gboolean
+accumulator_result (const RgAnalysisAcc * acc, gdouble * result_gain,
+    gdouble * result_peak)
+{
+  guint32 sum = 0;
+  guint32 upper;
+  guint i;
+
+  for (i = 0; i < G_N_ELEMENTS (acc->histogram); i++)
+    sum += acc->histogram[i];
+
+  if (sum == 0)
+    /* All entries are 0: We got less than 50ms of data. */
+    return FALSE;
+
+  upper = (guint32) ceil (sum * (1. - (gdouble) (RMS_PERCENTILE / 100.)));
+
+  for (i = G_N_ELEMENTS (acc->histogram); i--;) {
+    if (upper <= acc->histogram[i])
+      break;
+    upper -= acc->histogram[i];
+  }
+
+  if (result_peak != NULL)
+    *result_peak = acc->peak;
+  if (result_gain != NULL)
+    *result_gain = PINK_REF - (gdouble) i / STEPS_PER_DB;
+
+  return TRUE;
+}
+
+/* Functions that operate on contexts, for external usage. */
+
+/* Create a new context.  Before it can be used, a sample rate must be
+ * configured using rg_analysis_set_sample_rate. */
+
+RgAnalysisCtx *
+rg_analysis_new (void)
+{
+  RgAnalysisCtx *ctx;
+
+  ctx = g_new (RgAnalysisCtx, 1);
+
+  ctx->inpre_l = ctx->inprebuf_l + MAX_ORDER;
+  ctx->step_l = ctx->stepbuf_l + MAX_ORDER;
+  ctx->out_l = ctx->outbuf_l + MAX_ORDER;
+
+  ctx->inpre_r = ctx->inprebuf_r + MAX_ORDER;
+  ctx->step_r = ctx->stepbuf_r + MAX_ORDER;
+  ctx->out_r = ctx->outbuf_r + MAX_ORDER;
+
+  ctx->sample_rate = 0;
+
+  accumulator_clear (&ctx->track);
+  accumulator_clear (&ctx->album);
+
+  return ctx;
+}
+
+static void
+reset_silence_detection (RgAnalysisCtx * ctx)
+{
+  ctx->buffer_timestamp = GST_CLOCK_TIME_NONE;
+  ctx->buffer_n_samples_done = 0;
+}
+
+/* Adapt to given sample rate.  Does nothing if already the current
+ * rate (returns TRUE then).  Returns FALSE only if given sample rate
+ * is not supported.  If the configured rate changes, the last
+ * unprocessed incomplete 50ms chunk of data is dropped because the
+ * filters are reset. */
+
+gboolean
+rg_analysis_set_sample_rate (RgAnalysisCtx * ctx, gint sample_rate)
+{
+  g_return_val_if_fail (ctx != NULL, FALSE);
+
+  if (ctx->sample_rate == sample_rate)
+    return TRUE;
+
+  switch (sample_rate) {
+    case 48000:
+      ctx->sample_rate_index = 0;
+      break;
+    case 44100:
+      ctx->sample_rate_index = 1;
+      break;
+    case 32000:
+      ctx->sample_rate_index = 2;
+      break;
+    case 24000:
+      ctx->sample_rate_index = 3;
+      break;
+    case 22050:
+      ctx->sample_rate_index = 4;
+      break;
+    case 16000:
+      ctx->sample_rate_index = 5;
+      break;
+    case 12000:
+      ctx->sample_rate_index = 6;
+      break;
+    case 11025:
+      ctx->sample_rate_index = 7;
+      break;
+    case 8000:
+      ctx->sample_rate_index = 8;
+      break;
+    default:
+      return FALSE;
+  }
+
+  ctx->sample_rate = sample_rate;
+  /* The + 999 has the effect of ceil()ing: */
+  ctx->window_n_samples = (guint) ((sample_rate * RMS_WINDOW_MSECS + 999)
+      / 1000);
+
+  reset_filters (ctx);
+  reset_silence_detection (ctx);
+
+  return TRUE;
+}
+
+void
+rg_analysis_init_silence_detection (RgAnalysisCtx * ctx,
+    void (*post_message) (gpointer analysis, GstClockTime timestamp,
+        GstClockTime duration, gdouble rglevel), gpointer analysis)
+{
+  ctx->post_message = post_message;
+  ctx->analysis = analysis;
+  reset_silence_detection (ctx);
+}
+
+void
+rg_analysis_start_buffer (RgAnalysisCtx * ctx, GstClockTime buffer_timestamp)
+{
+  ctx->buffer_timestamp = buffer_timestamp;
+  ctx->buffer_n_samples_done = 0;
+}
+
+void
+rg_analysis_destroy (RgAnalysisCtx * ctx)
+{
+  g_free (ctx);
+}
+
+/* Entry points for analyzing sample data in common raw data formats.
+ * The stereo format functions expect interleaved frames.  It is
+ * possible to pass data in different formats for the same context,
+ * there are no restrictions.  All functions have the same signature;
+ * the depth argument for the float functions is not variable and must
+ * be given the value 32. */
+
+void
+rg_analysis_analyze_mono_float (RgAnalysisCtx * ctx, gconstpointer data,
+    gsize size, guint depth)
+{
+  gfloat conv_samples[512];
+  const gfloat *samples = (gfloat *) data;
+  guint n_samples = size / sizeof (gfloat);
+  gint i;
+
+  g_return_if_fail (depth == 32);
+  g_return_if_fail (size % sizeof (gfloat) == 0);
+
+  while (n_samples) {
+    gint n = MIN (n_samples, G_N_ELEMENTS (conv_samples));
+
+    n_samples -= n;
+    memcpy (conv_samples, samples, n * sizeof (gfloat));
+    for (i = 0; i < n; i++) {
+      ctx->track.peak = MAX (ctx->track.peak, fabs (conv_samples[i]));
+      conv_samples[i] *= 32768.;
+    }
+    samples += n;
+    rg_analysis_analyze (ctx, conv_samples, NULL, n);
+  }
+}
+
+void
+rg_analysis_analyze_stereo_float (RgAnalysisCtx * ctx, gconstpointer data,
+    gsize size, guint depth)
+{
+  gfloat conv_samples_l[256];
+  gfloat conv_samples_r[256];
+  const gfloat *samples = (gfloat *) data;
+  guint n_frames = size / (sizeof (gfloat) * 2);
+  gint i;
+
+  g_return_if_fail (depth == 32);
+  g_return_if_fail (size % (sizeof (gfloat) * 2) == 0);
+
+  while (n_frames) {
+    gint n = MIN (n_frames, G_N_ELEMENTS (conv_samples_l));
+
+    n_frames -= n;
+    for (i = 0; i < n; i++) {
+      gfloat old_sample;
+
+      old_sample = samples[2 * i];
+      ctx->track.peak = MAX (ctx->track.peak, fabs (old_sample));
+      conv_samples_l[i] = old_sample * 32768.;
+
+      old_sample = samples[2 * i + 1];
+      ctx->track.peak = MAX (ctx->track.peak, fabs (old_sample));
+      conv_samples_r[i] = old_sample * 32768.;
+    }
+    samples += 2 * n;
+    rg_analysis_analyze (ctx, conv_samples_l, conv_samples_r, n);
+  }
+}
+
+void
+rg_analysis_analyze_mono_int16 (RgAnalysisCtx * ctx, gconstpointer data,
+    gsize size, guint depth)
+{
+  gfloat conv_samples[512];
+  gint32 peak_sample = 0;
+  const gint16 *samples = (gint16 *) data;
+  guint n_samples = size / sizeof (gint16);
+  gint shift = sizeof (gint16) * 8 - depth;
+  gint i;
+
+  g_return_if_fail (depth <= (sizeof (gint16) * 8));
+  g_return_if_fail (size % sizeof (gint16) == 0);
+
+  while (n_samples) {
+    gint n = MIN (n_samples, G_N_ELEMENTS (conv_samples));
+
+    n_samples -= n;
+    for (i = 0; i < n; i++) {
+      gint16 old_sample = samples[i] << shift;
+
+      peak_sample = MAX (peak_sample, ABS ((gint32) old_sample));
+      conv_samples[i] = (gfloat) old_sample;
+    }
+    samples += n;
+    rg_analysis_analyze (ctx, conv_samples, NULL, n);
+  }
+  ctx->track.peak = MAX (ctx->track.peak,
+      (gdouble) peak_sample / ((gdouble) (1u << 15)));
+}
+
+void
+rg_analysis_analyze_stereo_int16 (RgAnalysisCtx * ctx, gconstpointer data,
+    gsize size, guint depth)
+{
+  gfloat conv_samples_l[256];
+  gfloat conv_samples_r[256];
+  gint32 peak_sample = 0;
+  const gint16 *samples = (gint16 *) data;
+  guint n_frames = size / (sizeof (gint16) * 2);
+  gint shift = sizeof (gint16) * 8 - depth;
+  gint i;
+
+  g_return_if_fail (depth <= (sizeof (gint16) * 8));
+  g_return_if_fail (size % (sizeof (gint16) * 2) == 0);
+
+  while (n_frames) {
+    gint n = MIN (n_frames, G_N_ELEMENTS (conv_samples_l));
+
+    n_frames -= n;
+    for (i = 0; i < n; i++) {
+      gint16 old_sample;
+
+      old_sample = samples[2 * i] << shift;
+      peak_sample = MAX (peak_sample, ABS ((gint32) old_sample));
+      conv_samples_l[i] = (gfloat) old_sample;
+
+      old_sample = samples[2 * i + 1] << shift;
+      peak_sample = MAX (peak_sample, ABS ((gint32) old_sample));
+      conv_samples_r[i] = (gfloat) old_sample;
+    }
+    samples += 2 * n;
+    rg_analysis_analyze (ctx, conv_samples_l, conv_samples_r, n);
+  }
+  ctx->track.peak = MAX (ctx->track.peak,
+      (gdouble) peak_sample / ((gdouble) (1u << 15)));
+}
+
+/* Analyze the given chunk of samples.  The sample data is given in
+ * floating point format but should be scaled such that the values
+ * +/-32768.0 correspond to the -0dBFS reference amplitude.
+ *
+ * samples_l: Buffer with sample data for the left channel or of the
+ * mono channel.
+ *
+ * samples_r: Buffer with sample data for the right channel or NULL
+ * for mono.
+ *
+ * n_samples: Number of samples passed in each buffer.
+ */
+
+void
+rg_analysis_analyze (RgAnalysisCtx * ctx, const gfloat * samples_l,
+    const gfloat * samples_r, guint n_samples)
+{
+  const gfloat *input_l, *input_r;
+  guint n_samples_done;
+  gint i;
+
+  g_return_if_fail (ctx != NULL);
+  g_return_if_fail (samples_l != NULL);
+  g_return_if_fail (ctx->sample_rate != 0);
+
+  if (n_samples == 0)
+    return;
+
+  if (samples_r == NULL)
+    /* Mono. */
+    samples_r = samples_l;
+
+  memcpy (ctx->inpre_l, samples_l,
+      MIN (n_samples, MAX_ORDER) * sizeof (gfloat));
+  memcpy (ctx->inpre_r, samples_r,
+      MIN (n_samples, MAX_ORDER) * sizeof (gfloat));
+
+  n_samples_done = 0;
+  while (n_samples_done < n_samples) {
+    /* Limit number of samples to be processed in this iteration to
+     * the number needed to complete the next window: */
+    guint n_samples_current = MIN (n_samples - n_samples_done,
+        ctx->window_n_samples - ctx->window_n_samples_done);
+
+    if (n_samples_done < MAX_ORDER) {
+      input_l = ctx->inpre_l + n_samples_done;
+      input_r = ctx->inpre_r + n_samples_done;
+      n_samples_current = MIN (n_samples_current, MAX_ORDER - n_samples_done);
+    } else {
+      input_l = samples_l + n_samples_done;
+      input_r = samples_r + n_samples_done;
+    }
+
+    apply_filters (ctx, input_l, input_r, n_samples_current);
+
+    /* Update the square sum. */
+    for (i = 0; i < n_samples_current; i++)
+      ctx->window_square_sum += ctx->out_l[ctx->window_n_samples_done + i]
+          * ctx->out_l[ctx->window_n_samples_done + i]
+          + ctx->out_r[ctx->window_n_samples_done + i]
+          * ctx->out_r[ctx->window_n_samples_done + i];
+
+    ctx->window_n_samples_done += n_samples_current;
+    ctx->buffer_n_samples_done += n_samples_current;
+
+    g_return_if_fail (ctx->window_n_samples_done <= ctx->window_n_samples);
+
+    if (ctx->window_n_samples_done == ctx->window_n_samples) {
+      /* Get the Root Mean Square (RMS) for this set of samples. */
+      gdouble val = STEPS_PER_DB * 10. * log10 (ctx->window_square_sum /
+          ctx->window_n_samples * 0.5 + 1.e-37);
+      gint ival = CLAMP ((gint) val, 0,
+          (gint) G_N_ELEMENTS (ctx->track.histogram) - 1);
+      /* Compute the per-window gain */
+      const gdouble gain = PINK_REF - (gdouble) ival / STEPS_PER_DB;
+      const GstClockTime timestamp = ctx->buffer_timestamp
+          + gst_util_uint64_scale_int_ceil (GST_SECOND,
+          ctx->buffer_n_samples_done,
+          ctx->sample_rate)
+          - RMS_WINDOW_MSECS * GST_MSECOND;
+
+      ctx->post_message (ctx->analysis, timestamp,
+          RMS_WINDOW_MSECS * GST_MSECOND, -gain);
+
+
+      ctx->track.histogram[ival]++;
+      ctx->window_square_sum = 0.;
+      ctx->window_n_samples_done = 0;
+
+      /* No need for memmove here, the areas never overlap: Even for
+       * the smallest sample rate, the number of samples needed for
+       * the window is greater than MAX_ORDER. */
+
+      memcpy (ctx->stepbuf_l, ctx->stepbuf_l + ctx->window_n_samples,
+          MAX_ORDER * sizeof (gfloat));
+      memcpy (ctx->outbuf_l, ctx->outbuf_l + ctx->window_n_samples,
+          MAX_ORDER * sizeof (gfloat));
+
+      memcpy (ctx->stepbuf_r, ctx->stepbuf_r + ctx->window_n_samples,
+          MAX_ORDER * sizeof (gfloat));
+      memcpy (ctx->outbuf_r, ctx->outbuf_r + ctx->window_n_samples,
+          MAX_ORDER * sizeof (gfloat));
+    }
+
+    n_samples_done += n_samples_current;
+  }
+
+  if (n_samples >= MAX_ORDER) {
+
+    memcpy (ctx->inprebuf_l, samples_l + n_samples - MAX_ORDER,
+        MAX_ORDER * sizeof (gfloat));
+
+    memcpy (ctx->inprebuf_r, samples_r + n_samples - MAX_ORDER,
+        MAX_ORDER * sizeof (gfloat));
+
+  } else {
+
+    memmove (ctx->inprebuf_l, ctx->inprebuf_l + n_samples,
+        (MAX_ORDER - n_samples) * sizeof (gfloat));
+    memcpy (ctx->inprebuf_l + MAX_ORDER - n_samples, samples_l,
+        n_samples * sizeof (gfloat));
+
+    memmove (ctx->inprebuf_r, ctx->inprebuf_r + n_samples,
+        (MAX_ORDER - n_samples) * sizeof (gfloat));
+    memcpy (ctx->inprebuf_r + MAX_ORDER - n_samples, samples_r,
+        n_samples * sizeof (gfloat));
+
+  }
+}
+
+/* Obtain track gain and peak.  Returns TRUE on success.  Can fail if
+ * not enough samples have been processed.  Updates album accumulator.
+ * Resets track accumulator. */
+
+gboolean
+rg_analysis_track_result (RgAnalysisCtx * ctx, gdouble * gain, gdouble * peak)
+{
+  gboolean result;
+
+  g_return_val_if_fail (ctx != NULL, FALSE);
+
+  accumulator_add (&ctx->album, &ctx->track);
+  result = accumulator_result (&ctx->track, gain, peak);
+  accumulator_clear (&ctx->track);
+
+  reset_filters (ctx);
+  reset_silence_detection (ctx);
+
+  return result;
+}
+
+/* Obtain album gain and peak.  Returns TRUE on success.  Can fail if
+ * not enough samples have been processed.  Resets album
+ * accumulator. */
+
+gboolean
+rg_analysis_album_result (RgAnalysisCtx * ctx, gdouble * gain, gdouble * peak)
+{
+  gboolean result;
+
+  g_return_val_if_fail (ctx != NULL, FALSE);
+
+  result = accumulator_result (&ctx->album, gain, peak);
+  accumulator_clear (&ctx->album);
+
+  return result;
+}
+
+void
+rg_analysis_reset_album (RgAnalysisCtx * ctx)
+{
+  accumulator_clear (&ctx->album);
+}
+
+/* Reset internal buffers as well as track and album accumulators.
+ * Configured sample rate is kept intact. */
+
+void
+rg_analysis_reset (RgAnalysisCtx * ctx)
+{
+  g_return_if_fail (ctx != NULL);
+
+  reset_filters (ctx);
+  accumulator_clear (&ctx->track);
+  accumulator_clear (&ctx->album);
+  reset_silence_detection (ctx);
+}
diff --git a/gst/replaygain/rganalysis.h b/gst/replaygain/rganalysis.h
new file mode 100644
index 0000000..f57ad0a
--- /dev/null
+++ b/gst/replaygain/rganalysis.h
@@ -0,0 +1,63 @@
+/* GStreamer ReplayGain analysis
+ *
+ * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
+ * Copyright (C) 2001 David Robinson <David@Robinson.org>
+ *                    Glen Sawyer <glensawyer@hotmail.com>
+ *
+ * rganalysis.h: Analyze raw audio data in accordance with ReplayGain
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 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
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifndef __RG_ANALYSIS_H__
+#define __RG_ANALYSIS_H__
+
+#include <glib.h>
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+typedef struct _RgAnalysisCtx RgAnalysisCtx;
+
+RgAnalysisCtx *rg_analysis_new (void);
+gboolean rg_analysis_set_sample_rate (RgAnalysisCtx * ctx, gint sample_rate);
+void rg_analysis_analyze_mono_float (RgAnalysisCtx * ctx, gconstpointer data,
+    gsize size, guint depth);
+void rg_analysis_analyze_stereo_float (RgAnalysisCtx * ctx, gconstpointer data,
+    gsize size, guint depth);
+void rg_analysis_analyze_mono_int16 (RgAnalysisCtx * ctx, gconstpointer data,
+    gsize size, guint depth);
+void rg_analysis_analyze_stereo_int16 (RgAnalysisCtx * ctx, gconstpointer data,
+    gsize size, guint depth);
+void rg_analysis_analyze (RgAnalysisCtx * ctx, const gfloat * samples_l,
+    const gfloat * samples_r, guint n_samples);
+gboolean rg_analysis_track_result (RgAnalysisCtx * ctx, gdouble * gain,
+    gdouble * peak);
+gboolean rg_analysis_album_result (RgAnalysisCtx * ctx, gdouble * gain,
+    gdouble * peak);
+void rg_analysis_init_silence_detection (
+    RgAnalysisCtx * ctx,
+    void (*post_message) (gpointer analysis, GstClockTime timestamp, GstClockTime duration, gdouble rglevel),
+    gpointer analysis);
+void rg_analysis_start_buffer (RgAnalysisCtx * ctx,
+                               GstClockTime buffer_timestamp);
+void rg_analysis_reset_album (RgAnalysisCtx * ctx);
+void rg_analysis_reset (RgAnalysisCtx * ctx);
+void rg_analysis_destroy (RgAnalysisCtx * ctx);
+
+G_END_DECLS
+
+#endif /* __RG_ANALYSIS_H__ */