diff -Naur gst-plugins-bad-orig/configure.ac gst-plugins-bad/configure.ac
--- gst-plugins-bad-orig/configure.ac	2006-09-15 08:19:36.000000000 +0200
+++ gst-plugins-bad/configure.ac	2006-09-30 15:33:03.000000000 +0200
@@ -82,6 +82,7 @@
   h264parse \
   modplug \
   nsf \
+  rganalysis \
   spectrum \
   speed \
   qtdemux \
@@ -845,6 +846,7 @@
 gst/modplug/Makefile
 gst/modplug/libmodplug/Makefile
 gst/nsf/Makefile
+gst/rganalysis/Makefile
 gst/spectrum/Makefile
 gst/speed/Makefile
 gst/qtdemux/Makefile
diff -Naur gst-plugins-bad-orig/docs/plugins/gst-plugins-bad-plugins-docs.sgml gst-plugins-bad/docs/plugins/gst-plugins-bad-plugins-docs.sgml
--- gst-plugins-bad-orig/docs/plugins/gst-plugins-bad-plugins-docs.sgml	2006-09-15 08:19:36.000000000 +0200
+++ gst-plugins-bad/docs/plugins/gst-plugins-bad-plugins-docs.sgml	2006-09-15 08:28:09.000000000 +0200
@@ -14,6 +14,7 @@
     <title>gst-plugins-bad Elements</title>
 
     <xi:include href="xml/element-dfbvideosink.xml" />
+    <xi:include href="xml/element-rganalysis.xml" />
     <xi:include href="xml/element-trm.xml" />
     <xi:include href="xml/element-videocrop.xml" />
     <xi:include href="xml/element-wavpackdec.xml" />
@@ -37,6 +38,7 @@
     <xi:include href="xml/plugin-musicbrainz.xml" />
     <xi:include href="xml/plugin-neon.xml" />
     <xi:include href="xml/plugin-qtdemux.xml" />
+    <xi:include href="xml/plugin-rganalysis.xml" />
     <xi:include href="xml/plugin-sdlvideosink.xml" />
     <xi:include href="xml/plugin-sdl.xml" />
     <xi:include href="xml/plugin-spectrum.xml" />
diff -Naur gst-plugins-bad-orig/docs/plugins/gst-plugins-bad-plugins-sections.txt gst-plugins-bad/docs/plugins/gst-plugins-bad-plugins-sections.txt
--- gst-plugins-bad-orig/docs/plugins/gst-plugins-bad-plugins-sections.txt	2006-09-15 08:19:36.000000000 +0200
+++ gst-plugins-bad/docs/plugins/gst-plugins-bad-plugins-sections.txt	2006-09-14 17:50:03.000000000 +0200
@@ -7,6 +7,14 @@
 </SECTION>
 
 <SECTION>
+<FILE>element-rganalysis</FILE>
+GstRgAnalysis
+<TITLE>rganalysis</TITLE>
+<SUBSECTION Standard>
+GstRgAnalysisClass
+</SECTION>
+
+<SECTION>
 <FILE>element-trm</FILE>
 GstTRM
 <TITLE>trm</TITLE>
diff -Naur gst-plugins-bad-orig/docs/plugins/Makefile.am gst-plugins-bad/docs/plugins/Makefile.am
--- gst-plugins-bad-orig/docs/plugins/Makefile.am	2006-09-15 08:19:36.000000000 +0200
+++ gst-plugins-bad/docs/plugins/Makefile.am	2006-09-20 03:09:28.000000000 +0200
@@ -91,6 +91,7 @@
 	$(top_srcdir)/ext/wavpack/gstwavpackdec.h \
 	$(top_srcdir)/ext/wavpack/gstwavpackenc.h \
 	$(top_srcdir)/ext/wavpack/gstwavpackparse.h \
+	$(top_srcdir)/gst/rganalysis/gstrganalysis.h \
 	$(top_srcdir)/gst/videocrop/gstvideocrop.h
 
 # Images to copy into HTML directory.
diff -Naur gst-plugins-bad-orig/gst/rganalysis/gstrganalysis.c gst-plugins-bad/gst/rganalysis/gstrganalysis.c
--- gst-plugins-bad-orig/gst/rganalysis/gstrganalysis.c	1970-01-01 01:00:00.000000000 +0100
+++ gst-plugins-bad/gst/rganalysis/gstrganalysis.c	2006-09-21 17:29:54.000000000 +0200
@@ -0,0 +1,686 @@
+/* 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
+ *
+ * <refsect2>
+ * <para>
+ * GstRgAnalysis 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 and #GST_TAG_TRACK_PEAK.
+ * </para>
+ * <title>Album processing</title>
+ * <para>
+ * 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 receive
+ * the same amplification.  This keeps the relative volume levels
+ * between the tracks intact.  To enable this, set the <link
+ * linkend="GstRgAnalysis--num-tracks">num-tracks</link> 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
+ * will be 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 make sure that the
+ * album gain and peak values are also associated with the other
+ * tracks when storing the results.  It is thus a bit more complex to
+ * implement, but should not be avoided since the album gain is
+ * generally more valuable for use during playback than the track
+ * gain.
+ * </para>
+ * <title>Skipping processing</title> 
+ * <para>
+ * For assisting transcoder/converter applications, the element can
+ * silently skip the processing of streams that already contain the
+ * necessary meta data 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 the <link
+ * linkend="GstRgAnalysis--forced">forced</link> property to #FALSE.
+ * If used in conjunction with album processing, 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, your
+ * application has to scan all files beforehand and enable forcing of
+ * processing if needed.
+ * </para>
+ * <title>Tips</title>
+ * <itemizedlist>
+ * <listitem><para>
+ * 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 since the album gain and peak values need to be
+ * associated with all tracks of an album, not just the last one.
+ * </para></listitem>
+ * <listitem><para>
+ * 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 rganalysis element's state using
+ * gst_element_set_locked_state() when it is in PAUSED or PLAYING.  As
+ * with any other element, don't forget to unlock it again and set it
+ * to the NULL state before dropping the last reference.
+ * </para></listitem>
+ * <listitem><para>
+ * If the total number of album tracks is unknown beforehand, set the
+ * num-tracks property to some large value like #G_MAXINT (or set it
+ * to >= 2 before each track starts).  Before the last track ends, set
+ * the property value to 1.
+ * </para></listitem>
+ * </itemizedlist>
+ * <title>Compliance</title>
+ * <para>
+ * Analyzing the ReplayGain pink noise reference waveform will compute
+ * a result of +6.00dB instead of the expected 0.00dB because the
+ * default reference level is 89dB.  To obtain values as lined out in
+ * the original proposal of ReplayGain, set the <link
+ * linkend="GstRgAnalysis--reference-level">reference-level</link>
+ * property to 83.  Almost all software uses 89dB as a reference
+ * however, which works against the tendency of the algorithm to
+ * advise to drastically lower the volume of music with a highly
+ * compressed dynamic range and high average output levels.  This
+ * tendency is normally to be fought during playback (if wanted), by
+ * using a default pre-amp value of at least +6.00dB.  At one point,
+ * the majority of analyzer implementations switched to 89dB which
+ * moved this adjustment to the analyzing/metadata writing process.
+ * This change has been acknowledged by the author of the ReplayGain
+ * proposal, however at the time of this writing, the webpage is still
+ * not updated.
+ * </para>
+ * <title>Example launch lines</title>
+ * <para>Analyze a simple test waveform:</para>
+ * <programlisting>
+ * gst-launch -t audiotestsrc wave=sine num-buffers=512 ! rganalysis ! fakesink
+ * </programlisting>
+ * <para>Analyze a given file:</para>
+ * <programlisting>
+ * gst-launch -t filesrc location="Some file.ogg" ! decodebin ! audioconvert ! audioresample ! rganalysis ! fakesink
+ * </programlisting>
+ * <para>Analyze the pink noise reference file:</para>
+ * <programlisting>
+ * gst-launch -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav ! wavparse ! rganalysis ! fakesink
+ * </programlisting>
+ * <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 <string.h>
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+
+#include "gstrganalysis.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_rg_analysis_debug);
+#define GST_CAT_DEFAULT gst_rg_analysis_debug
+
+static const GstElementDetails rganalysis_details = {
+  "ReplayGain analysis",
+  "Filter/Analyzer/Audio",
+  "Perform the ReplayGain analysis",
+  "Ren\xc3\xa9 Stadler <mail@renestadler.de>"
+};
+
+/* Default property value. */
+#define FORCED_DEFAULT TRUE
+
+enum
+{
+  PROP_0,
+  PROP_NUM_TRACKS,
+  PROP_FORCED,
+  PROP_REFERENCE_LEVEL
+};
+
+/* 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                                                \
+  "channels = (int) { 1, 2 }, "                                         \
+  "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 ("audio/x-raw-float, "
+        "width = (int) 32, " "endianness = (int) BYTE_ORDER, "
+        REPLAY_GAIN_CAPS "; "
+        "audio/x-raw-int, "
+        "width = (int) 16, " "depth = (int) [ 1, 16 ], "
+        "signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
+        REPLAY_GAIN_CAPS));
+
+static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
+        "width = (int) 32, " "endianness = (int) BYTE_ORDER, "
+        REPLAY_GAIN_CAPS "; "
+        "audio/x-raw-int, "
+        "width = (int) 16, " "depth = (int) [ 1, 16 ], "
+        "signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
+        REPLAY_GAIN_CAPS));
+
+GST_BOILERPLATE (GstRgAnalysis, gst_rg_analysis, GstBaseTransform,
+    GST_TYPE_BASE_TRANSFORM);
+
+static void gst_rg_analysis_class_init (GstRgAnalysisClass * klass);
+static void gst_rg_analysis_init (GstRgAnalysis * filter,
+    GstRgAnalysisClass * gclass);
+
+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_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_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
+
+  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_details (element_class, &rganalysis_details);
+
+  GST_DEBUG_CATEGORY_INIT (gst_rg_analysis_debug, "rganalysis", 0,
+      "ReplayGain analysis element");
+}
+
+static void
+gst_rg_analysis_class_init (GstRgAnalysisClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstBaseTransformClass *trans_class;
+
+  gobject_class = (GObjectClass *) klass;
+  gobject_class->set_property = gst_rg_analysis_set_property;
+  gobject_class->get_property = gst_rg_analysis_get_property;
+
+  g_object_class_install_property (gobject_class, PROP_NUM_TRACKS,
+      g_param_spec_int ("num-tracks", "Number of album tracks",
+          "Number of remaining tracks in the album",
+          0, G_MAXINT, 0, G_PARAM_READWRITE));
+  g_object_class_install_property (gobject_class, PROP_FORCED,
+      g_param_spec_boolean ("forced", "Force processing",
+          "Analyze streams even when ReplayGain tags exist",
+          FORCED_DEFAULT, G_PARAM_READWRITE));
+  g_object_class_install_property (gobject_class, PROP_REFERENCE_LEVEL,
+      g_param_spec_double ("reference-level", "Reference level",
+          "Reference level in dB (83.0 for original proposal)",
+          0.0, G_MAXDOUBLE, RG_REFERENCE_LEVEL, G_PARAM_READWRITE));
+
+  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->event = GST_DEBUG_FUNCPTR (gst_rg_analysis_event);
+  trans_class->stop = GST_DEBUG_FUNCPTR (gst_rg_analysis_stop);
+  trans_class->passthrough_on_same_caps = TRUE;
+}
+
+static void
+gst_rg_analysis_init (GstRgAnalysis * filter, GstRgAnalysisClass * gclass)
+{
+  filter->num_tracks = 0;
+  filter->forced = FORCED_DEFAULT;
+  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);
+
+  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;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rg_analysis_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRgAnalysis *filter = GST_RG_ANALYSIS (object);
+
+  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;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+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 ();
+  filter->analyze = NULL;
+
+  GST_DEBUG_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);
+  GstStructure *structure;
+  const gchar *mime_type;
+  gint n_channels, sample_rate, sample_bit_size, sample_size;
+
+  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);
+
+  structure = gst_caps_get_structure (in_caps, 0);
+  mime_type = gst_structure_get_name (structure);
+
+  if (!gst_structure_get_int (structure, "width", &sample_bit_size)
+      || !gst_structure_get_int (structure, "channels", &n_channels)
+      || !gst_structure_get_int (structure, "rate", &sample_rate))
+    goto invalid_format;
+
+  if (!rg_analysis_set_sample_rate (filter->ctx, sample_rate))
+    goto invalid_format;
+
+  if (sample_bit_size % 8 != 0)
+    goto invalid_format;
+  sample_size = sample_bit_size / 8;
+
+  if (strcmp (mime_type, "audio/x-raw-float") == 0) {
+
+    if (sample_size != sizeof (gfloat))
+      goto invalid_format;
+
+    /* 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 (n_channels == 1)
+      filter->analyze = rg_analysis_analyze_mono_float;
+    else if (n_channels == 2)
+      filter->analyze = rg_analysis_analyze_stereo_float;
+    else
+      goto invalid_format;
+
+  } else if (strcmp (mime_type, "audio/x-raw-int") == 0) {
+
+    if (sample_size != sizeof (gint16))
+      goto invalid_format;
+
+    if (!gst_structure_get_int (structure, "depth", &filter->depth))
+      goto invalid_format;
+    if (filter->depth < 1 || filter->depth > 16)
+      goto invalid_format;
+
+    if (n_channels == 1)
+      filter->analyze = rg_analysis_analyze_mono_int16;
+    else if (n_channels == 2)
+      filter->analyze = rg_analysis_analyze_stereo_int16;
+    else
+      goto invalid_format;
+
+  } else {
+
+    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);
+
+  g_return_val_if_fail (filter->ctx != NULL, GST_FLOW_ERROR);
+  g_return_val_if_fail (filter->analyze != NULL, GST_FLOW_ERROR);
+
+  if (filter->skip)
+    return GST_FLOW_OK;
+
+  GST_DEBUG_OBJECT (filter, "Processing buffer of size %u",
+      GST_BUFFER_SIZE (buf));
+
+  filter->analyze (filter->ctx, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
+      filter->depth);
+
+  return GST_FLOW_OK;
+}
+
+static gboolean
+gst_rg_analysis_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_DEBUG_OBJECT (filter, "Received EOS event");
+
+      gst_rg_analysis_handle_eos (filter);
+
+      GST_DEBUG_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 TRUE;
+}
+
+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_DEBUG_OBJECT (filter, "Stopped");
+
+  return TRUE;
+}
+
+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_INFO_OBJECT (filter, "Ignoring TAG event: Skipping album");
+    return;
+  } else if (filter->skip) {
+    GST_INFO_OBJECT (filter, "Ignoring TAG event: Skipping track");
+    return;
+  } else if (filter->ignore_tags) {
+    GST_INFO_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_INFO_OBJECT (filter, "Track tags not complete yet");
+    return;
+  }
+
+  if (album_processing && !(filter->has_album_gain && filter->has_album_peak)) {
+    GST_INFO_OBJECT (filter, "Album tags not complete yet");
+    return;
+  }
+
+  if (filter->forced) {
+    GST_INFO_OBJECT (filter,
+        "Existing tags are sufficient, but processing anyway (forced)");
+    return;
+  }
+
+  filter->skip = TRUE;
+  rg_analysis_reset (filter->ctx);
+
+  if (!album_processing)
+    GST_INFO_OBJECT (filter,
+        "Existing tags are sufficient, will not process this track");
+  else
+    GST_INFO_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_DEBUG_OBJECT (filter, "Posting tag list with results");
+      /* This steals our reference to the list: */
+      gst_element_found_tags_for_pad (GST_ELEMENT (filter),
+          GST_BASE_TRANSFORM_SRC_PAD (GST_BASE_TRANSFORM (filter)), tag_list);
+    }
+  }
+
+  if (album_processing) {
+    filter->num_tracks--;
+
+    if (!album_finished)
+      GST_INFO_OBJECT (filter, "Album not finished yet (num-tracks is now %u)",
+          filter->num_tracks);
+    else
+      GST_INFO_OBJECT (filter, "Album finished (num-tracks is now 0)");
+  }
+
+  if (album_processing)
+    g_object_notify (G_OBJECT (filter), "num-tracks");
+}
+
+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 ();
+    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 ();
+    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;
+}
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  return gst_element_register (plugin, "rganalysis", GST_RANK_NONE,
+      GST_TYPE_RG_ANALYSIS);
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "rganalysis",
+    "ReplayGain analysis", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME,
+    GST_PACKAGE_ORIGIN);
diff -Naur gst-plugins-bad-orig/gst/rganalysis/gstrganalysis.h gst-plugins-bad/gst/rganalysis/gstrganalysis.h
--- gst-plugins-bad-orig/gst/rganalysis/gstrganalysis.h	1970-01-01 01:00:00.000000000 +0100
+++ gst-plugins-bad/gst/rganalysis/gstrganalysis.h	2006-09-30 14:58:34.000000000 +0200
@@ -0,0 +1,83 @@
+/* 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;
+
+  /* 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;
+};
+
+G_END_DECLS
+
+#endif /* __GST_RG_ANALYSIS_H__ */
diff -Naur gst-plugins-bad-orig/gst/rganalysis/Makefile.am gst-plugins-bad/gst/rganalysis/Makefile.am
--- gst-plugins-bad-orig/gst/rganalysis/Makefile.am	1970-01-01 01:00:00.000000000 +0100
+++ gst-plugins-bad/gst/rganalysis/Makefile.am	2006-09-20 20:00:31.000000000 +0200
@@ -0,0 +1,13 @@
+plugin_LTLIBRARIES = libgstrganalysis.la
+
+libgstrganalysis_la_SOURCES = \
+	gstrganalysis.c       \
+	rganalysis.c
+libgstrganalysis_la_CFLAGS  = $(GST_CFLAGS) $(GST_BASE_CFLAGS)
+libgstrganalysis_la_LIBADD  = $(GST_LIBS) $(GST_BASE_LIBS)
+libgstrganalysis_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
+
+# headers we need but don't want installed
+noinst_HEADERS =         \
+	gstrganalysis.h  \
+	rganalysis.h
diff -Naur gst-plugins-bad-orig/gst/rganalysis/rganalysis.c gst-plugins-bad/gst/rganalysis/rganalysis.c
--- gst-plugins-bad-orig/gst/rganalysis/rganalysis.c	1970-01-01 01:00:00.000000000 +0100
+++ gst-plugins-bad/gst/rganalysis/rganalysis.c	2006-09-30 18:07:15.000000000 +0200
@@ -0,0 +1,772 @@
+/* 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;
+};
+
+/* 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 G_OS_WIN32
+/* Disable double-to-float warning: */
+#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 G_OS_WIN32
+#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)
+{
+  output[0] = 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;
+}
+
+/* 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);
+
+  return TRUE;
+}
+
+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;
+
+    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);
+
+      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);
+
+  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);
+}
diff -Naur gst-plugins-bad-orig/gst/rganalysis/rganalysis.h gst-plugins-bad/gst/rganalysis/rganalysis.h
--- gst-plugins-bad-orig/gst/rganalysis/rganalysis.h	1970-01-01 01:00:00.000000000 +0100
+++ gst-plugins-bad/gst/rganalysis/rganalysis.h	2006-09-21 02:58:29.000000000 +0200
@@ -0,0 +1,58 @@
+/* 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>
+
+G_BEGIN_DECLS
+
+#define RG_REFERENCE_LEVEL 89.
+
+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_reset_album (RgAnalysisCtx * ctx);
+void rg_analysis_reset (RgAnalysisCtx * ctx);
+void rg_analysis_destroy (RgAnalysisCtx * ctx);
+
+G_END_DECLS
+
+#endif /* __RG_ANALYSIS_H__ */
diff -Naur gst-plugins-bad-orig/tests/check/elements/rganalysis.c gst-plugins-bad/tests/check/elements/rganalysis.c
--- gst-plugins-bad-orig/tests/check/elements/rganalysis.c	1970-01-01 01:00:00.000000000 +0100
+++ gst-plugins-bad/tests/check/elements/rganalysis.c	2006-09-21 17:29:54.000000000 +0200
@@ -0,0 +1,1871 @@
+/* GStreamer ReplayGain analysis
+ *
+ * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
+ *
+ * rganalysis.c: Unit test for the rganalysis element
+ *
+ * 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
+ */
+
+/* Some things to note about the RMS window length of the analysis
+ * algorithm and thus the implementation used in the element:
+ * Processing divides input data into 50ms windows at some point.
+ * Some details about this that normally do not matter:
+ *
+ *  1. At the end of a stream, the remainder of data that did not fill
+ *     up the last 50ms window is simply discarded.
+ *
+ *  2. If the sample rate changes during a stream, the currently
+ *     running window is discarded and the equal loudness filter gets
+ *     reset as if a new stream started.
+ *
+ *  3. For the album gain, it is not entirely correct to think of
+ *     obtaining it like "as if all the tracks are analyzed as one
+ *     track".  There isn't a separate window being tracked for album
+ *     processing, so at stream (track) end, the remaining unfilled
+ *     window does not contribute to the album gain either.
+ *
+ *  4. If a waveform with a result gain G is concatenated to itself
+ *     and the result processed as a track, the gain can be different
+ *     from G if and only if the duration of the original waveform is
+ *     not an integer multiple of 50ms.  If the original waveform gets
+ *     processed as a single track and then the same data again as a
+ *     subsequent track, the album result gain will always match G
+ *     (this is implied by 3.).
+ *
+ *  5. A stream shorter than 50ms cannot be analyzed.  At 8000 and
+ *     48000 Hz, this corresponds to 400 resp. 2400 frames.  If a
+ *     stream is shorter than 50ms, the element will not generate tags
+ *     at EOS (only if an album finished, but only album tags are
+ *     generated then).  This is not an erroneous condition, the
+ *     element should behave normally.
+ *
+ * The limitations outlined in 1.-4. do not apply to the peak values.
+ * Every single sample is accounted for when looking for the peak.
+ * Thus the album peak is guaranteed to be the maximum value of all
+ * track peaks.
+ *
+ * In normal day-to-day use, these little facts are unlikely to be
+ * relevant, but they have to be kept in mind for writing the tests
+ * here.
+ */
+
+#include <gst/check/gstcheck.h>
+
+GList *buffers = NULL;
+
+/* For ease of programming we use globals to keep refs for our floating
+ * src and sink pads we create; otherwise we always have to do get_pad,
+ * get_peer, and then remove references in every test function */
+static GstPad *mysrcpad, *mysinkpad;
+
+/* Mapping from supported sample rates to the correct result gain for
+ * the following test waveform: 20 * 512 samples with a quarter-full
+ * amplitude of toggling sign, changing every 48 samples and starting
+ * with the positive value.
+ *
+ * Even if we would generate a wave describing a signal with the same
+ * frequency at each sampling rate, the results would vary (slightly).
+ * Hence the simple generation method, since we cannot use a constant
+ * value as expected result anyways.  For all sample rates, changing
+ * the sign every 48 frames gives a sane frequency.  Buffers
+ * containing data that forms such a waveform is created using the
+ * test_buffer_square_{float,int16}_{mono,stereo} functions below.
+ *
+ * The results have been checked against what the metaflac and
+ * wavegain programs generate for such a stream.  If you want to
+ * verify these, be sure that the metaflac program does not produce
+ * incorrect results in your environment: I found a strange bug in the
+ * (defacto) reference code for the analysis that sometimes leads to
+ * incorrect RMS window lengths. */
+
+struct rate_test
+{
+  guint sample_rate;
+  gdouble gain;
+};
+
+static const struct rate_test supported_rates[] = {
+  8000, -0.91,
+  11025, -2.80,
+  12000, -3.13,
+  16000, -4.26,
+  22050, -5.64,
+  24000, -5.87,
+  32000, -6.03,
+  44100, -6.20,
+  48000, -6.14
+};
+
+/* Lookup the correct gain adjustment result in above array. */
+
+static gdouble
+get_expected_gain (guint sample_rate)
+{
+  gint i;
+
+  for (i = G_N_ELEMENTS (supported_rates); i--;)
+    if (supported_rates[i].sample_rate == sample_rate)
+      return supported_rates[i].gain;
+  g_return_val_if_reached (0.0);
+}
+
+#define SILENCE_GAIN 64.82
+
+#define REPLAY_GAIN_CAPS                                \
+  "channels = (int) { 1, 2 }, "                         \
+  "rate = (int) { 8000, 11025, 12000, 16000, 22050, "   \
+  "24000, 32000, 44100, 48000 }"
+
+#define RG_ANALYSIS_CAPS_TEMPLATE_STRING  \
+  "audio/x-raw-float, "                   \
+  "width = (int) 32, "                    \
+  "endianness = (int) BYTE_ORDER, "       \
+  REPLAY_GAIN_CAPS                        \
+  "; "                                    \
+  "audio/x-raw-int, "                     \
+  "width = (int) 16, "                    \
+  "depth = (int) [ 1, 16 ], "             \
+  "signed = (boolean) true, "             \
+  "endianness = (int) BYTE_ORDER, "       \
+  REPLAY_GAIN_CAPS
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (RG_ANALYSIS_CAPS_TEMPLATE_STRING)
+    );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (RG_ANALYSIS_CAPS_TEMPLATE_STRING)
+    );
+
+GstElement *
+setup_rganalysis ()
+{
+  GstElement *analysis;
+  GstBus *bus;
+
+  GST_DEBUG ("setup_rganalysis");
+  analysis = gst_check_setup_element ("rganalysis");
+  mysrcpad = gst_check_setup_src_pad (analysis, &srctemplate, NULL);
+  mysinkpad = gst_check_setup_sink_pad (analysis, &sinktemplate, NULL);
+  gst_pad_set_active (mysrcpad, TRUE);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  bus = gst_bus_new ();
+  gst_element_set_bus (analysis, bus);
+  /* gst_element_set_bus does not steal a reference. */
+  gst_object_unref (bus);
+
+  return analysis;
+}
+
+void
+cleanup_rganalysis (GstElement * element)
+{
+  GST_DEBUG ("cleanup_rganalysis");
+
+  g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
+  g_list_free (buffers);
+  buffers = NULL;
+
+  /* The bus owns references to the element: */
+  gst_element_set_bus (element, NULL);
+
+  gst_check_teardown_src_pad (element);
+  gst_check_teardown_sink_pad (element);
+  gst_check_teardown_element (element);
+}
+
+static void
+set_playing_state (GstElement * element)
+{
+  fail_unless (gst_element_set_state (element,
+          GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
+      "Could not set state to PLAYING");
+}
+
+static void
+send_eos_event (GstElement * element)
+{
+  GstBus *bus = gst_element_get_bus (element);
+  GstPad *pad = gst_element_get_pad (element, "sink");
+  GstEvent *event = gst_event_new_eos ();
+
+  fail_unless (gst_pad_send_event (pad, event),
+      "Cannot send EOS event: Not handled.");
+
+  /* There is no sink element, so _we_ post the EOS message on the bus
+   * here.  Of course we generate any EOS ourselves, but this allows
+   * us to poll for the EOS message in poll_eos if we expect the
+   * element to _not_ generate a TAG message.  That's better than
+   * waiting for a timeout to lapse. */
+  fail_unless (gst_bus_post (bus, gst_message_new_eos (NULL)));
+
+  gst_object_unref (bus);
+  gst_object_unref (pad);
+}
+
+static void
+send_tag_event (GstElement * element, GstTagList * tag_list)
+{
+  GstPad *pad = gst_element_get_pad (element, "sink");
+  GstEvent *event = gst_event_new_tag (tag_list);
+
+  fail_unless (gst_pad_send_event (pad, event),
+      "Cannot send TAG event: Not handled.");
+
+  gst_object_unref (pad);
+}
+
+static void
+poll_eos (GstElement * element)
+{
+  GstBus *bus = gst_element_get_bus (element);
+  GstMessage *message;
+
+  message = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_TAG, GST_SECOND);
+  fail_unless (message != NULL, "Could not poll for EOS message: Timed out");
+  fail_unless (message->type == GST_MESSAGE_EOS,
+      "Could not poll for eos message: got message of type %s instead",
+      gst_message_type_get_name (message->type));
+
+  gst_message_unref (message);
+  gst_object_unref (bus);
+}
+
+/* This also polls for EOS since the TAG message comes right before
+ * the end of streams. */
+
+static GstTagList *
+poll_tags (GstElement * element)
+{
+  GstBus *bus = gst_element_get_bus (element);
+  GstTagList *tag_list;
+  GstMessage *message;
+
+  message = gst_bus_poll (bus, GST_MESSAGE_TAG, GST_SECOND);
+  fail_unless (message != NULL, "Could not poll for TAG message: Timed out");
+
+  fail_unless (GST_MESSAGE_SRC (message) == GST_OBJECT (element));
+
+  gst_message_parse_tag (message, &tag_list);
+  gst_message_unref (message);
+  gst_object_unref (bus);
+
+  poll_eos (element);
+
+  return tag_list;
+}
+
+#define MATCH_PEAK(p1, p2) ((p1 < p2 + 1e-6) && (p2 < p1 + 1e-6))
+#define MATCH_GAIN(g1, g2) ((g1 < g2 + 1e-13) && (g2 < g1 + 1e-13))
+
+static void
+fail_unless_track_gain (const GstTagList * tag_list, gdouble gain)
+{
+  gdouble result;
+
+  fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN, &result),
+      "Tag list contains no track gain value");
+  fail_unless (MATCH_GAIN (gain, result),
+      "Track gain %+.2f does not match, expected %+.2f", result, gain);
+}
+
+static void
+fail_unless_track_peak (const GstTagList * tag_list, gdouble peak)
+{
+  gdouble result;
+
+  fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK, &result),
+      "Tag list contains no track peak value");
+  fail_unless (MATCH_PEAK (peak, result),
+      "Track peak %f does not match, expected %f", result, peak);
+}
+
+static void
+fail_unless_album_gain (const GstTagList * tag_list, gdouble gain)
+{
+  gdouble result;
+
+  fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN, &result),
+      "Tag list contains no album gain value");
+  fail_unless (MATCH_GAIN (result, gain),
+      "Album gain %+.2f does not match, expected %+.2f", result, gain);
+}
+
+static void
+fail_unless_album_peak (const GstTagList * tag_list, gdouble peak)
+{
+  gdouble result;
+
+  fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK, &result),
+      "Tag list contains no album peak value");
+  fail_unless (MATCH_PEAK (peak, result),
+      "Album peak %f does not match, expected %f", result, peak);
+}
+
+static void
+fail_if_track_tags (const GstTagList * tag_list)
+{
+  gdouble result;
+
+  fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN, &result),
+      "Tag list contains track gain value (but should not)");
+  fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK, &result),
+      "Tag list contains track peak value (but should not)");
+}
+
+static void
+fail_if_album_tags (const GstTagList * tag_list)
+{
+  gdouble result;
+
+  fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN, &result),
+      "Tag list contains album gain value (but should not)");
+  fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK, &result),
+      "Tag list contains album peak value (but should not)");
+}
+
+static void
+fail_unless_num_tracks (GstElement * element, guint num_tracks)
+{
+  guint current;
+
+  g_object_get (element, "num-tracks", &current, NULL);
+  fail_unless (current == num_tracks,
+      "num-tracks property has incorrect value %u, expected %u",
+      current, num_tracks);
+}
+
+/* Functions that create buffers with constant sample values, for peak
+ * tests. */
+
+static GstBuffer *
+test_buffer_const_float_mono (gint sample_rate, gsize n_frames, gfloat value)
+{
+  GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat));
+  gfloat *data = (gfloat *) GST_BUFFER_DATA (buf);
+  GstCaps *caps;
+  gint i;
+
+  for (i = n_frames; i--;)
+    *data++ = value;
+
+  caps = gst_caps_new_simple ("audio/x-raw-float",
+      "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1,
+      "endianess", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL);
+  gst_buffer_set_caps (buf, caps);
+  gst_caps_unref (caps);
+
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
+
+  return buf;
+}
+
+static GstBuffer *
+test_buffer_const_float_stereo (gint sample_rate, gsize n_frames,
+    gfloat value_l, gfloat value_r)
+{
+  GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat) * 2);
+  gfloat *data = (gfloat *) GST_BUFFER_DATA (buf);
+  GstCaps *caps;
+  gint i;
+
+  for (i = n_frames; i--;) {
+    *data++ = value_l;
+    *data++ = value_r;
+  }
+
+  caps = gst_caps_new_simple ("audio/x-raw-float",
+      "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2,
+      "endianess", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL);
+  gst_buffer_set_caps (buf, caps);
+  gst_caps_unref (caps);
+
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
+
+  return buf;
+}
+
+static GstBuffer *
+test_buffer_const_int16_mono (gint sample_rate, gint depth, gsize n_frames,
+    gint16 value)
+{
+  GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16));
+  gint16 *data = (gint16 *) GST_BUFFER_DATA (buf);
+  GstCaps *caps;
+  gint i;
+
+  for (i = n_frames; i--;)
+    *data++ = value;
+
+  caps = gst_caps_new_simple ("audio/x-raw-int",
+      "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1,
+      "endianess", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE,
+      "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL);
+  gst_buffer_set_caps (buf, caps);
+  gst_caps_unref (caps);
+
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
+
+  return buf;
+}
+
+static GstBuffer *
+test_buffer_const_int16_stereo (gint sample_rate, gint depth, gsize n_frames,
+    gint16 value_l, gint16 value_r)
+{
+  GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16) * 2);
+  gint16 *data = (gint16 *) GST_BUFFER_DATA (buf);
+  GstCaps *caps;
+  gint i;
+
+  for (i = n_frames; i--;) {
+    *data++ = value_l;
+    *data++ = value_r;
+  }
+
+  caps = gst_caps_new_simple ("audio/x-raw-int",
+      "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2,
+      "endianess", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE,
+      "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL);
+  gst_buffer_set_caps (buf, caps);
+  gst_caps_unref (caps);
+
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
+
+  return buf;
+}
+
+/* Functions that create data buffers containing square signal
+ * waveforms. */
+
+static GstBuffer *
+test_buffer_square_float_mono (gint * accumulator, gint sample_rate,
+    gsize n_frames, gfloat value)
+{
+  GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat));
+  gfloat *data = (gfloat *) GST_BUFFER_DATA (buf);
+  GstCaps *caps;
+  gint i;
+
+  for (i = n_frames; i--;) {
+    *accumulator += 1;
+    *accumulator %= 96;
+
+    if (*accumulator < 48)
+      *data++ = value;
+    else
+      *data++ = -value;
+  }
+
+  caps = gst_caps_new_simple ("audio/x-raw-float",
+      "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1,
+      "endianess", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL);
+  gst_buffer_set_caps (buf, caps);
+  gst_caps_unref (caps);
+
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
+
+  return buf;
+}
+
+static GstBuffer *
+test_buffer_square_float_stereo (gint * accumulator, gint sample_rate,
+    gsize n_frames, gfloat value_l, gfloat value_r)
+{
+  GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat) * 2);
+  gfloat *data = (gfloat *) GST_BUFFER_DATA (buf);
+  GstCaps *caps;
+  gint i;
+
+  for (i = n_frames; i--;) {
+    *accumulator += 1;
+    *accumulator %= 96;
+
+    if (*accumulator < 48) {
+      *data++ = value_l;
+      *data++ = value_r;
+    } else {
+      *data++ = -value_l;
+      *data++ = -value_r;
+    }
+  }
+
+  caps = gst_caps_new_simple ("audio/x-raw-float",
+      "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2,
+      "endianess", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL);
+  gst_buffer_set_caps (buf, caps);
+  gst_caps_unref (caps);
+
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
+
+  return buf;
+}
+
+static GstBuffer *
+test_buffer_square_int16_mono (gint * accumulator, gint sample_rate,
+    gint depth, gsize n_frames, gint16 value)
+{
+  GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16));
+  gint16 *data = (gint16 *) GST_BUFFER_DATA (buf);
+  GstCaps *caps;
+  gint i;
+
+  for (i = n_frames; i--;) {
+    *accumulator += 1;
+    *accumulator %= 96;
+
+    if (*accumulator < 48)
+      *data++ = value;
+    else
+      *data++ = -MAX (value, -32767);
+  }
+
+  caps = gst_caps_new_simple ("audio/x-raw-int",
+      "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1,
+      "endianess", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE,
+      "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL);
+  gst_buffer_set_caps (buf, caps);
+  gst_caps_unref (caps);
+
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
+
+  return buf;
+}
+
+static GstBuffer *
+test_buffer_square_int16_stereo (gint * accumulator, gint sample_rate,
+    gint depth, gsize n_frames, gint16 value_l, gint16 value_r)
+{
+  GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16) * 2);
+  gint16 *data = (gint16 *) GST_BUFFER_DATA (buf);
+  GstCaps *caps;
+  gint i;
+
+  for (i = n_frames; i--;) {
+    *accumulator += 1;
+    *accumulator %= 96;
+
+    if (*accumulator < 48) {
+      *data++ = value_l;
+      *data++ = value_r;
+    } else {
+      *data++ = -MAX (value_l, -32767);
+      *data++ = -MAX (value_r, -32767);
+    }
+  }
+
+  caps = gst_caps_new_simple ("audio/x-raw-int",
+      "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2,
+      "endianess", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE,
+      "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL);
+  gst_buffer_set_caps (buf, caps);
+  gst_caps_unref (caps);
+
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
+
+  return buf;
+}
+
+static void
+push_buffer (GstBuffer * buf)
+{
+  /* gst_pad_push steals a reference. */
+  fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
+}
+
+/*** Start of the tests. ***/
+
+/* This test looks redundant, but early versions of the element
+ * crashed when doing, well, nothing: */
+
+GST_START_TEST (test_no_buffer)
+{
+  GstElement *element = setup_rganalysis ();
+
+  set_playing_state (element);
+  send_eos_event (element);
+  poll_eos (element);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_no_buffer_album_1)
+{
+  GstElement *element = setup_rganalysis ();
+
+  set_playing_state (element);
+
+  /* Single track: */
+  send_eos_event (element);
+  poll_eos (element);
+
+  /* First album: */
+  g_object_set (element, "num-tracks", 3, NULL);
+
+  send_eos_event (element);
+  poll_eos (element);
+  fail_unless_num_tracks (element, 2);
+
+  send_eos_event (element);
+  poll_eos (element);
+  fail_unless_num_tracks (element, 1);
+
+  send_eos_event (element);
+  poll_eos (element);
+  fail_unless_num_tracks (element, 0);
+
+  /* Second album: */
+  g_object_set (element, "num-tracks", 2, NULL);
+
+  send_eos_event (element);
+  poll_eos (element);
+  fail_unless_num_tracks (element, 1);
+
+  send_eos_event (element);
+  poll_eos (element);
+  fail_unless_num_tracks (element, 0);
+
+  /* Single track: */
+  send_eos_event (element);
+  poll_eos (element);
+  fail_unless_num_tracks (element, 0);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_no_buffer_album_2)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  gint accumulator = 0;
+  gint i;
+
+  g_object_set (element, "num-tracks", 3, NULL);
+  set_playing_state (element);
+
+  /* No buffer for the first track. */
+
+  send_eos_event (element);
+  /* No tags should be posted, there was nothing to analyze: */
+  poll_eos (element);
+  fail_unless_num_tracks (element, 2);
+
+  /* A test waveform with known gain result as second track: */
+
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_mono (&accumulator, 44100, 512,
+            0.25));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, -6.20);
+  /* Album is not finished yet: */
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 1);
+
+  /* No buffer for the last track. */
+
+  send_eos_event (element);
+
+  tag_list = poll_tags (element);
+  fail_unless_album_peak (tag_list, 0.25);
+  fail_unless_album_gain (tag_list, -6.20);
+  /* No track tags should be posted, as there was no data for it: */
+  fail_if_track_tags (tag_list);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 0);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_empty_buffers)
+{
+  GstElement *element = setup_rganalysis ();
+
+  set_playing_state (element);
+
+  /* Single track: */
+  push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0));
+  send_eos_event (element);
+  poll_eos (element);
+
+  /* First album: */
+  g_object_set (element, "num-tracks", 2, NULL);
+
+  push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0));
+  send_eos_event (element);
+  poll_eos (element);
+  fail_unless_num_tracks (element, 1);
+
+  push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0));
+  send_eos_event (element);
+  poll_eos (element);
+  fail_unless_num_tracks (element, 0);
+
+  /* Second album, with a single track: */
+  g_object_set (element, "num-tracks", 1, NULL);
+  push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0));
+  send_eos_event (element);
+  poll_eos (element);
+  fail_unless_num_tracks (element, 0);
+
+  /* Single track: */
+  push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0));
+  send_eos_event (element);
+  poll_eos (element);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+/* Tests for correctness of the peak values. */
+
+/* Float peak test.  For stereo, one channel has the constant value of
+ * -1.369, the other one 0.0.  This tests many things: The result peak
+ * value should occur on any channel.  The peak is of course the
+ * absolute amplitude, so 1.369 should be the result.  This will also
+ * detect if the code uses the absolute value during the comparison.
+ * If it is buggy it will return 0.0 since 0.0 > -1.369.  Furthermore,
+ * this makes sure that there is no problem with headroom (exceeding
+ * 0dBFS).  In the wild you get float samples > 1.0 from stuff like
+ * vorbis. */
+
+GST_START_TEST (test_peak_float)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+
+  set_playing_state (element);
+  push_buffer (test_buffer_const_float_stereo (8000, 512, -1.369, 0.0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.369);
+  gst_tag_list_free (tag_list);
+
+  /* Swapped channels. */
+  push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, -1.369));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.369);
+  gst_tag_list_free (tag_list);
+
+  /* Mono. */
+  push_buffer (test_buffer_const_float_mono (8000, 512, -1.369));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.369);
+  gst_tag_list_free (tag_list);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_peak_int16_16)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+
+  set_playing_state (element);
+
+  /* Half amplitude. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 1 << 14, 0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+  /* Swapped channels. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, 1 << 14));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+  /* Mono. */
+  push_buffer (test_buffer_const_int16_mono (8000, 16, 512, 1 << 14));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+  /* Half amplitude, negative variant. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -1 << 14, 0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+  /* Swapped channels. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -1 << 14));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+  /* Mono. */
+  push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -1 << 14));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+
+  /* Now check for correct normalization of the peak value: Sample
+   * values of this format range from -32768 to 32767.  So for the
+   * highest positive amplitude we do not reach 1.0, only for
+   * -32768! */
+
+  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 32767, 0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 32767. / 32768.);
+  gst_tag_list_free (tag_list);
+
+  /* Swapped channels. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, 32767));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 32767. / 32768.);
+  gst_tag_list_free (tag_list);
+
+  /* Mono. */
+  push_buffer (test_buffer_const_int16_mono (8000, 16, 512, 32767));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 32767. / 32768.);
+  gst_tag_list_free (tag_list);
+
+
+  /* Negative variant, reaching 1.0. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -32768, 0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.0);
+  gst_tag_list_free (tag_list);
+
+  /* Swapped channels. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -32768));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.0);
+  gst_tag_list_free (tag_list);
+
+  /* Mono. */
+  push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -32768));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.0);
+  gst_tag_list_free (tag_list);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+/* Same as the test before, but with 8 bits (packed into 16 bits). */
+
+GST_START_TEST (test_peak_int16_8)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+
+  set_playing_state (element);
+
+  /* Half amplitude. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 1 << 6, 0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+  /* Swapped channels. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, 1 << 6));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+  /* Mono. */
+  push_buffer (test_buffer_const_int16_mono (8000, 8, 512, 1 << 6));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+
+  /* Half amplitude, negative variant. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, -1 << 6, 0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+  /* Swapped channels. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, -1 << 6));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+  /* Mono. */
+  push_buffer (test_buffer_const_int16_mono (8000, 8, 512, -1 << 6));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+
+
+  /* Almost full amplitude (maximum positive value). */
+  push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, (1 << 7) - 1, 0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.9921875);
+  gst_tag_list_free (tag_list);
+
+  /* Swapped channels. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, (1 << 7) - 1));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.9921875);
+  gst_tag_list_free (tag_list);
+
+  /* Mono. */
+  push_buffer (test_buffer_const_int16_mono (8000, 8, 512, (1 << 7) - 1));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.9921875);
+  gst_tag_list_free (tag_list);
+
+
+  /* Full amplitude (maximum negative value). */
+  push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, -1 << 7, 0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.0);
+  gst_tag_list_free (tag_list);
+
+  /* Swapped channels. */
+  push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, -1 << 7));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.0);
+  gst_tag_list_free (tag_list);
+
+  /* Mono. */
+  push_buffer (test_buffer_const_int16_mono (8000, 8, 512, -1 << 7));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.0);
+  gst_tag_list_free (tag_list);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_peak_album)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+
+  g_object_set (element, "num-tracks", 2, NULL);
+  set_playing_state (element);
+
+  push_buffer (test_buffer_const_float_stereo (8000, 1024, 1.0, 0.0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.0);
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 1);
+
+  push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.0, 0.5));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  fail_unless_album_peak (tag_list, 1.0);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 0);
+
+  /* Try a second album: */
+  g_object_set (element, "num-tracks", 3, NULL);
+
+  push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.4, 0.4));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.4);
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 2);
+
+  push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.45, 0.45));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.45);
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 1);
+
+  push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.2, 0.2));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.2);
+  fail_unless_album_peak (tag_list, 0.45);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 0);
+
+  /* And now a single track, not in album mode (num-tracks is 0
+   * now): */
+  push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.1, 0.1));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.1);
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+/* Switching from track to album mode. */
+
+GST_START_TEST (test_peak_track_album)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  guint num;
+
+  set_playing_state (element);
+
+  push_buffer (test_buffer_const_float_mono (8000, 1024, 1.0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.0);
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+
+  g_object_set (element, "num-tracks", 1, NULL);
+  push_buffer (test_buffer_const_float_mono (8000, 1024, 0.5));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  fail_unless_album_peak (tag_list, 0.5);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 0);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+/* Disabling album processing before the end of the album.  Probably a
+ * rare edge case and applications should not rely on this to work.
+ * They need to send the element to the READY state to clear up after
+ * an aborted album anyway since they might need to process another
+ * album afterwards. */
+
+GST_START_TEST (test_peak_album_abort_to_track)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  guint num;
+
+  g_object_set (element, "num-tracks", 2, NULL);
+  set_playing_state (element);
+
+  push_buffer (test_buffer_const_float_stereo (8000, 1024, 1.0, 0.0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 1.0);
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 1);
+
+  g_object_set (element, "num-tracks", 0, NULL);
+
+  push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.0, 0.5));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_gain_album)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  gint accumulator;
+  gint i;
+
+  g_object_set (element, "num-tracks", 3, NULL);
+  set_playing_state (element);
+
+  /* The three tracks are constructed such that if any of these is in
+   * fact ignored for the album gain, the album gain will differ. */
+
+  accumulator = 0;
+  for (i = 8; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
+            0.75, 0.75));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.75);
+  fail_unless_track_gain (tag_list, -15.70);
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+
+  accumulator = 0;
+  for (i = 12; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
+            0.5, 0.5));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.5);
+  fail_unless_track_gain (tag_list, -12.22);
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+
+  accumulator = 0;
+  for (i = 180; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
+            0.25, 0.25));
+  send_eos_event (element);
+
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, -6.20);
+  fail_unless_album_peak (tag_list, 0.75);
+  /* Strangely, wavegain reports -12.17 for the album, but the fixed
+   * metaflac agrees to us.  Could be a 32767 vs. 32768 issue. */
+  fail_unless_album_gain (tag_list, -12.18);
+  gst_tag_list_free (tag_list);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+/* Checks ensuring that the "forced" property works as advertised. */
+
+GST_START_TEST (test_forced)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  gint accumulator = 0;
+  gint i;
+
+  g_object_set (element, "forced", FALSE, NULL);
+  set_playing_state (element);
+
+  tag_list = gst_tag_list_new ();
+  /* Provided values are totally arbitrary. */
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
+      GST_TAG_TRACK_PEAK, 1.0, GST_TAG_TRACK_GAIN, 2.21, NULL);
+  send_tag_event (element, tag_list);
+
+  for (i = 20; i--;)
+    push_buffer (test_buffer_const_float_stereo (44100, 512, 0.5, 0.5));
+  send_eos_event (element);
+  /* This fails if a tag message is generated: */
+  poll_eos (element);
+
+  /* Now back to a track without tags. */
+
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
+            0.25, 0.25));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, get_expected_gain (44100));
+  gst_tag_list_free (tag_list);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+/* Sending track gain and peak in separate tag lists. */
+
+GST_START_TEST (test_forced_separate)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  gint accumulator = 0;
+  gint i;
+
+  g_object_set (element, "forced", FALSE, NULL);
+  set_playing_state (element);
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, GST_TAG_TRACK_GAIN, 2.21,
+      NULL);
+  send_tag_event (element, tag_list);
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, GST_TAG_TRACK_PEAK, 1.0,
+      NULL);
+  send_tag_event (element, tag_list);
+
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
+            0.5, 0.5));
+  send_eos_event (element);
+  /* This fails if a tag message is generated: */
+  poll_eos (element);
+
+  /* Now a track without tags. */
+
+  accumulator = 0;
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
+            0.25, 0.25));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, get_expected_gain (44100));
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+/* A TAG event is sent _after_ data has already been processed.  In
+ * real pipelines, this could happen if there is more than one
+ * rganalysis element (by accident).  While it would have analyzed all
+ * the data prior to receiving the event, I expect it to not post its
+ * results if not forced.  This test is almost equivalent to
+ * test_forced. */
+
+GST_START_TEST (test_forced_after_data)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  gint accumulator = 0;
+  gint i;
+
+  g_object_set (element, "forced", FALSE, NULL);
+  set_playing_state (element);
+
+  for (i = 20; i--;)
+    push_buffer (test_buffer_const_float_stereo (8000, 512, 0.5, 0.5));
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
+      GST_TAG_TRACK_PEAK, 1.0, GST_TAG_TRACK_GAIN, 2.21, NULL);
+  send_tag_event (element, tag_list);
+
+  send_eos_event (element);
+  poll_eos (element);
+
+  /* Now back to a normal track, this one has no tags: */
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25,
+            0.25));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, get_expected_gain (8000));
+  gst_tag_list_free (tag_list);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+/* Like test_forced, but *analyze* an album afterwards.  The two tests
+ * following this one check the *skipping* of albums. */
+
+GST_START_TEST (test_forced_album)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  gint accumulator;
+  gint i;
+
+  g_object_set (element, "forced", FALSE, NULL);
+  set_playing_state (element);
+
+  tag_list = gst_tag_list_new ();
+  /* Provided values are totally arbitrary. */
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
+      GST_TAG_TRACK_PEAK, 1.0, GST_TAG_TRACK_GAIN, 2.21, NULL);
+  send_tag_event (element, tag_list);
+
+  accumulator = 0;
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
+            0.5, 0.5));
+  send_eos_event (element);
+  /* This fails if a tag message is generated: */
+  poll_eos (element);
+
+  /* Now an album without tags. */
+  g_object_set (element, "num-tracks", 2, NULL);
+
+  accumulator = 0;
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
+            0.25, 0.25));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, get_expected_gain (44100));
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 1);
+
+  accumulator = 0;
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
+            0.25, 0.25));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, get_expected_gain (44100));
+  fail_unless_album_peak (tag_list, 0.25);
+  fail_unless_album_gain (tag_list, get_expected_gain (44100));
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 0);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_forced_album_skip)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  gint accumulator = 0;
+  gint i;
+
+  g_object_set (element, "forced", FALSE, "num-tracks", 2, NULL);
+  set_playing_state (element);
+
+  tag_list = gst_tag_list_new ();
+  /* Provided values are totally arbitrary. */
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
+      GST_TAG_TRACK_PEAK, 0.75, GST_TAG_TRACK_GAIN, 2.21,
+      GST_TAG_ALBUM_PEAK, 0.80, GST_TAG_ALBUM_GAIN, -0.11, NULL);
+  send_tag_event (element, tag_list);
+
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25,
+            0.25));
+  send_eos_event (element);
+  poll_eos (element);
+  fail_unless_num_tracks (element, 1);
+
+  /* This track has no tags, but needs to be skipped anyways since we
+   * are in album processing mode. */
+  for (i = 20; i--;)
+    push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, 0.0));
+  send_eos_event (element);
+  poll_eos (element);
+  fail_unless_num_tracks (element, 0);
+
+  /* Normal track after the album.  Of course not to be skipped. */
+  accumulator = 0;
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25,
+            0.25));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, get_expected_gain (8000));
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_forced_album_no_skip)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  gint accumulator = 0;
+  gint i;
+
+  g_object_set (element, "forced", FALSE, "num-tracks", 2, NULL);
+  set_playing_state (element);
+
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25,
+            0.25));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, get_expected_gain (8000));
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 1);
+
+  /* The second track has indeed full tags, but although being not
+   * forced, this one has to be processed because album processing is
+   * on. */
+  tag_list = gst_tag_list_new ();
+  /* Provided values are totally arbitrary. */
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
+      GST_TAG_TRACK_PEAK, 0.75, GST_TAG_TRACK_GAIN, 2.21,
+      GST_TAG_ALBUM_PEAK, 0.80, GST_TAG_ALBUM_GAIN, -0.11, NULL);
+  send_tag_event (element, tag_list);
+  for (i = 20; i--;)
+    push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, 0.0));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.0);
+  fail_unless_track_gain (tag_list, SILENCE_GAIN);
+  /* Second track was just silence so the album peak equals the first
+   * track's peak. */
+  fail_unless_album_peak (tag_list, 0.25);
+  /* Statistical processing leads to the second track being
+   * ignored for the gain (because it is so short): */
+  fail_unless_album_gain (tag_list, get_expected_gain (8000));
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 0);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_forced_abort_album_no_skip)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  gint accumulator = 0;
+  gint i;
+
+  g_object_set (element, "forced", FALSE, "num-tracks", 2, NULL);
+  set_playing_state (element);
+
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25,
+            0.25));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, get_expected_gain (8000));
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+  fail_unless_num_tracks (element, 1);
+
+  /* Disabling album processing before end of album: */
+  g_object_set (element, "num-tracks", 0, NULL);
+
+  /* Processing a track that has to be skipped. */
+  tag_list = gst_tag_list_new ();
+  /* Provided values are totally arbitrary. */
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
+      GST_TAG_TRACK_PEAK, 0.75, GST_TAG_TRACK_GAIN, 2.21,
+      GST_TAG_ALBUM_PEAK, 0.80, GST_TAG_ALBUM_GAIN, -0.11, NULL);
+  send_tag_event (element, tag_list);
+  for (i = 20; i--;)
+    push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, 0.0));
+  send_eos_event (element);
+  poll_eos (element);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_reference_level)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  gint accumulator = 0;
+  gint i;
+
+  g_object_set (element, "reference-level", 83., "num-tracks", 2, NULL);
+  set_playing_state (element);
+
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
+            0.25, 0.25));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, get_expected_gain (44100) - 6.);
+  fail_if_album_tags (tag_list);
+  gst_tag_list_free (tag_list);
+
+  accumulator = 0;
+  for (i = 20; i--;)
+    push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
+            0.25, 0.25));
+  send_eos_event (element);
+  tag_list = poll_tags (element);
+  fail_unless_track_peak (tag_list, 0.25);
+  fail_unless_track_gain (tag_list, get_expected_gain (44100) - 6.);
+  fail_unless_album_peak (tag_list, 0.25);
+  /* We provided the same waveform twice, with a reset separating
+   * them.  Therefore, the album gain matches the track gain. */
+  fail_unless_album_gain (tag_list, get_expected_gain (44100) - 6.);
+  gst_tag_list_free (tag_list);
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_all_formats)
+{
+  GstElement *element = setup_rganalysis ();
+  GstTagList *tag_list;
+  gint accumulator = 0;
+  gint i, j;
+
+  set_playing_state (element);
+  for (i = G_N_ELEMENTS (supported_rates); i--;) {
+    accumulator = 0;
+    for (j = 0; j < 4; j++)
+      push_buffer (test_buffer_square_float_stereo (&accumulator,
+              supported_rates[i].sample_rate, 512, 0.25, 0.25));
+    for (j = 0; j < 3; j++)
+      push_buffer (test_buffer_square_float_mono (&accumulator,
+              supported_rates[i].sample_rate, 512, 0.25));
+    for (j = 0; j < 4; j++)
+      push_buffer (test_buffer_square_int16_stereo (&accumulator,
+              supported_rates[i].sample_rate, 16, 512, 1 << 13, 1 << 13));
+    for (j = 0; j < 3; j++)
+      push_buffer (test_buffer_square_int16_mono (&accumulator,
+              supported_rates[i].sample_rate, 16, 512, 1 << 13));
+    for (j = 0; j < 3; j++)
+      push_buffer (test_buffer_square_int16_stereo (&accumulator,
+              supported_rates[i].sample_rate, 8, 512, 1 << 5, 1 << 5));
+    for (j = 0; j < 3; j++)
+      push_buffer (test_buffer_square_int16_mono (&accumulator,
+              supported_rates[i].sample_rate, 8, 512, 1 << 5));
+    send_eos_event (element);
+    tag_list = poll_tags (element);
+    fail_unless_track_peak (tag_list, 0.25);
+    fail_unless_track_gain (tag_list, supported_rates[i].gain);
+    gst_tag_list_free (tag_list);
+  }
+
+  cleanup_rganalysis (element);
+}
+
+GST_END_TEST;
+
+/* Checks ensuring all advertised supported sample rates are really
+ * accepted, for integer and float, mono and stereo.  This also
+ * verifies that the correct gain is computed for all formats (except
+ * odd bit depths). */
+
+#define MAKE_GAIN_TEST_FLOAT_MONO(sample_rate)                        \
+  GST_START_TEST (test_gain_float_mono_##sample_rate)                 \
+{                                                                     \
+  GstElement *element = setup_rganalysis ();                          \
+  GstTagList *tag_list;                                               \
+  gint accumulator = 0;                                               \
+  gint i;                                                             \
+                                                                      \
+  set_playing_state (element);                                        \
+                                                                      \
+  for (i = 0; i < 20; i++)                                            \
+    push_buffer (test_buffer_square_float_mono (&accumulator,         \
+            sample_rate, 512, 0.25));                                 \
+  send_eos_event (element);                                           \
+  tag_list = poll_tags (element);                                     \
+  fail_unless_track_peak (tag_list, 0.25);                            \
+  fail_unless_track_gain (tag_list,                                   \
+      get_expected_gain (sample_rate));                               \
+  gst_tag_list_free (tag_list);                                       \
+                                                                      \
+  cleanup_rganalysis (element);                                       \
+}                                                                     \
+                                                                      \
+GST_END_TEST;
+
+#define MAKE_GAIN_TEST_FLOAT_STEREO(sample_rate)                      \
+  GST_START_TEST (test_gain_float_stereo_##sample_rate)               \
+{                                                                     \
+  GstElement *element = setup_rganalysis ();                          \
+  GstTagList *tag_list;                                               \
+  gint accumulator = 0;                                               \
+  gint i;                                                             \
+                                                                      \
+  set_playing_state (element);                                        \
+                                                                      \
+  for (i = 0; i < 20; i++)                                            \
+    push_buffer (test_buffer_square_float_stereo (&accumulator,       \
+            sample_rate, 512, 0.25, 0.25));                           \
+  send_eos_event (element);                                           \
+  tag_list = poll_tags (element);                                     \
+  fail_unless_track_peak (tag_list, 0.25);                            \
+  fail_unless_track_gain (tag_list,                                   \
+      get_expected_gain (sample_rate));                               \
+  gst_tag_list_free (tag_list);                                       \
+                                                                      \
+  cleanup_rganalysis (element);                                       \
+}                                                                     \
+                                                                      \
+GST_END_TEST;
+
+#define MAKE_GAIN_TEST_INT16_MONO(sample_rate, depth)                 \
+  GST_START_TEST (test_gain_int16_##depth##_mono_##sample_rate)       \
+{                                                                     \
+  GstElement *element = setup_rganalysis ();                          \
+  GstTagList *tag_list;                                               \
+  gint accumulator = 0;                                               \
+  gint i;                                                             \
+                                                                      \
+  set_playing_state (element);                                        \
+                                                                      \
+  for (i = 0; i < 20; i++)                                            \
+    push_buffer (test_buffer_square_int16_mono (&accumulator,         \
+            sample_rate, depth, 512, 1 << (13 + depth - 16)));        \
+                                                                      \
+  send_eos_event (element);                                           \
+  tag_list = poll_tags (element);                                     \
+  fail_unless_track_peak (tag_list, 0.25);                            \
+  fail_unless_track_gain (tag_list,                                   \
+      get_expected_gain (sample_rate));                               \
+  gst_tag_list_free (tag_list);                                       \
+                                                                      \
+  cleanup_rganalysis (element);                                       \
+}                                                                     \
+                                                                      \
+GST_END_TEST;
+
+#define MAKE_GAIN_TEST_INT16_STEREO(sample_rate, depth)               \
+  GST_START_TEST (test_gain_int16_##depth##_stereo_##sample_rate)     \
+{                                                                     \
+  GstElement *element = setup_rganalysis ();                          \
+  GstTagList *tag_list;                                               \
+  gint accumulator = 0;                                               \
+  gint i;                                                             \
+                                                                      \
+  set_playing_state (element);                                        \
+                                                                      \
+  for (i = 0; i < 20; i++)                                            \
+    push_buffer (test_buffer_square_int16_stereo (&accumulator,       \
+            sample_rate, depth, 512, 1 << (13 + depth - 16),          \
+            1 << (13 + depth - 16)));                                 \
+  send_eos_event (element);                                           \
+  tag_list = poll_tags (element);                                     \
+  fail_unless_track_peak (tag_list, 0.25);                            \
+  fail_unless_track_gain (tag_list,                                   \
+      get_expected_gain (sample_rate));                               \
+  gst_tag_list_free (tag_list);                                       \
+                                                                      \
+  cleanup_rganalysis (element);                                       \
+}                                                                     \
+                                                                      \
+GST_END_TEST;
+
+MAKE_GAIN_TEST_FLOAT_MONO (8000);
+MAKE_GAIN_TEST_FLOAT_MONO (11025);
+MAKE_GAIN_TEST_FLOAT_MONO (12000);
+MAKE_GAIN_TEST_FLOAT_MONO (16000);
+MAKE_GAIN_TEST_FLOAT_MONO (22050);
+MAKE_GAIN_TEST_FLOAT_MONO (24000);
+MAKE_GAIN_TEST_FLOAT_MONO (32000);
+MAKE_GAIN_TEST_FLOAT_MONO (44100);
+MAKE_GAIN_TEST_FLOAT_MONO (48000);
+
+MAKE_GAIN_TEST_FLOAT_STEREO (8000);
+MAKE_GAIN_TEST_FLOAT_STEREO (11025);
+MAKE_GAIN_TEST_FLOAT_STEREO (12000);
+MAKE_GAIN_TEST_FLOAT_STEREO (16000);
+MAKE_GAIN_TEST_FLOAT_STEREO (22050);
+MAKE_GAIN_TEST_FLOAT_STEREO (24000);
+MAKE_GAIN_TEST_FLOAT_STEREO (32000);
+MAKE_GAIN_TEST_FLOAT_STEREO (44100);
+MAKE_GAIN_TEST_FLOAT_STEREO (48000);
+
+MAKE_GAIN_TEST_INT16_MONO (8000, 16);
+MAKE_GAIN_TEST_INT16_MONO (11025, 16);
+MAKE_GAIN_TEST_INT16_MONO (12000, 16);
+MAKE_GAIN_TEST_INT16_MONO (16000, 16);
+MAKE_GAIN_TEST_INT16_MONO (22050, 16);
+MAKE_GAIN_TEST_INT16_MONO (24000, 16);
+MAKE_GAIN_TEST_INT16_MONO (32000, 16);
+MAKE_GAIN_TEST_INT16_MONO (44100, 16);
+MAKE_GAIN_TEST_INT16_MONO (48000, 16);
+
+MAKE_GAIN_TEST_INT16_STEREO (8000, 16);
+MAKE_GAIN_TEST_INT16_STEREO (11025, 16);
+MAKE_GAIN_TEST_INT16_STEREO (12000, 16);
+MAKE_GAIN_TEST_INT16_STEREO (16000, 16);
+MAKE_GAIN_TEST_INT16_STEREO (22050, 16);
+MAKE_GAIN_TEST_INT16_STEREO (24000, 16);
+MAKE_GAIN_TEST_INT16_STEREO (32000, 16);
+MAKE_GAIN_TEST_INT16_STEREO (44100, 16);
+MAKE_GAIN_TEST_INT16_STEREO (48000, 16);
+
+MAKE_GAIN_TEST_INT16_MONO (8000, 8);
+MAKE_GAIN_TEST_INT16_MONO (11025, 8);
+MAKE_GAIN_TEST_INT16_MONO (12000, 8);
+MAKE_GAIN_TEST_INT16_MONO (16000, 8);
+MAKE_GAIN_TEST_INT16_MONO (22050, 8);
+MAKE_GAIN_TEST_INT16_MONO (24000, 8);
+MAKE_GAIN_TEST_INT16_MONO (32000, 8);
+MAKE_GAIN_TEST_INT16_MONO (44100, 8);
+MAKE_GAIN_TEST_INT16_MONO (48000, 8);
+
+MAKE_GAIN_TEST_INT16_STEREO (8000, 8);
+MAKE_GAIN_TEST_INT16_STEREO (11025, 8);
+MAKE_GAIN_TEST_INT16_STEREO (12000, 8);
+MAKE_GAIN_TEST_INT16_STEREO (16000, 8);
+MAKE_GAIN_TEST_INT16_STEREO (22050, 8);
+MAKE_GAIN_TEST_INT16_STEREO (24000, 8);
+MAKE_GAIN_TEST_INT16_STEREO (32000, 8);
+MAKE_GAIN_TEST_INT16_STEREO (44100, 8);
+MAKE_GAIN_TEST_INT16_STEREO (48000, 8);
+
+Suite *
+rganalysis_suite (void)
+{
+  Suite *s = suite_create ("rganalysis");
+  TCase *tc_chain = tcase_create ("general");
+
+  suite_add_tcase (s, tc_chain);
+
+  tcase_add_test (tc_chain, test_no_buffer);
+  tcase_add_test (tc_chain, test_no_buffer_album_1);
+  tcase_add_test (tc_chain, test_no_buffer_album_2);
+  tcase_add_test (tc_chain, test_empty_buffers);
+
+  tcase_add_test (tc_chain, test_peak_float);
+  tcase_add_test (tc_chain, test_peak_int16_16);
+  tcase_add_test (tc_chain, test_peak_int16_8);
+
+  tcase_add_test (tc_chain, test_peak_album);
+  tcase_add_test (tc_chain, test_peak_track_album);
+  tcase_add_test (tc_chain, test_peak_album_abort_to_track);
+
+  tcase_add_test (tc_chain, test_gain_album);
+
+  tcase_add_test (tc_chain, test_forced);
+  tcase_add_test (tc_chain, test_forced_separate);
+  tcase_add_test (tc_chain, test_forced_after_data);
+  tcase_add_test (tc_chain, test_forced_album);
+  tcase_add_test (tc_chain, test_forced_album_skip);
+  tcase_add_test (tc_chain, test_forced_album_no_skip);
+  tcase_add_test (tc_chain, test_forced_abort_album_no_skip);
+
+  tcase_add_test (tc_chain, test_reference_level);
+
+  tcase_add_test (tc_chain, test_all_formats);
+
+  tcase_add_test (tc_chain, test_gain_float_mono_8000);
+  tcase_add_test (tc_chain, test_gain_float_mono_11025);
+  tcase_add_test (tc_chain, test_gain_float_mono_12000);
+  tcase_add_test (tc_chain, test_gain_float_mono_16000);
+  tcase_add_test (tc_chain, test_gain_float_mono_22050);
+  tcase_add_test (tc_chain, test_gain_float_mono_24000);
+  tcase_add_test (tc_chain, test_gain_float_mono_32000);
+  tcase_add_test (tc_chain, test_gain_float_mono_44100);
+  tcase_add_test (tc_chain, test_gain_float_mono_48000);
+
+  tcase_add_test (tc_chain, test_gain_float_stereo_8000);
+  tcase_add_test (tc_chain, test_gain_float_stereo_11025);
+  tcase_add_test (tc_chain, test_gain_float_stereo_12000);
+  tcase_add_test (tc_chain, test_gain_float_stereo_16000);
+  tcase_add_test (tc_chain, test_gain_float_stereo_22050);
+  tcase_add_test (tc_chain, test_gain_float_stereo_24000);
+  tcase_add_test (tc_chain, test_gain_float_stereo_32000);
+  tcase_add_test (tc_chain, test_gain_float_stereo_44100);
+  tcase_add_test (tc_chain, test_gain_float_stereo_48000);
+
+  tcase_add_test (tc_chain, test_gain_int16_16_mono_8000);
+  tcase_add_test (tc_chain, test_gain_int16_16_mono_11025);
+  tcase_add_test (tc_chain, test_gain_int16_16_mono_12000);
+  tcase_add_test (tc_chain, test_gain_int16_16_mono_16000);
+  tcase_add_test (tc_chain, test_gain_int16_16_mono_22050);
+  tcase_add_test (tc_chain, test_gain_int16_16_mono_24000);
+  tcase_add_test (tc_chain, test_gain_int16_16_mono_32000);
+  tcase_add_test (tc_chain, test_gain_int16_16_mono_44100);
+  tcase_add_test (tc_chain, test_gain_int16_16_mono_48000);
+
+  tcase_add_test (tc_chain, test_gain_int16_16_stereo_8000);
+  tcase_add_test (tc_chain, test_gain_int16_16_stereo_11025);
+  tcase_add_test (tc_chain, test_gain_int16_16_stereo_12000);
+  tcase_add_test (tc_chain, test_gain_int16_16_stereo_16000);
+  tcase_add_test (tc_chain, test_gain_int16_16_stereo_22050);
+  tcase_add_test (tc_chain, test_gain_int16_16_stereo_24000);
+  tcase_add_test (tc_chain, test_gain_int16_16_stereo_32000);
+  tcase_add_test (tc_chain, test_gain_int16_16_stereo_44100);
+  tcase_add_test (tc_chain, test_gain_int16_16_stereo_48000);
+
+  tcase_add_test (tc_chain, test_gain_int16_8_mono_8000);
+  tcase_add_test (tc_chain, test_gain_int16_8_mono_11025);
+  tcase_add_test (tc_chain, test_gain_int16_8_mono_12000);
+  tcase_add_test (tc_chain, test_gain_int16_8_mono_16000);
+  tcase_add_test (tc_chain, test_gain_int16_8_mono_22050);
+  tcase_add_test (tc_chain, test_gain_int16_8_mono_24000);
+  tcase_add_test (tc_chain, test_gain_int16_8_mono_32000);
+  tcase_add_test (tc_chain, test_gain_int16_8_mono_44100);
+  tcase_add_test (tc_chain, test_gain_int16_8_mono_48000);
+
+  tcase_add_test (tc_chain, test_gain_int16_8_stereo_8000);
+  tcase_add_test (tc_chain, test_gain_int16_8_stereo_11025);
+  tcase_add_test (tc_chain, test_gain_int16_8_stereo_12000);
+  tcase_add_test (tc_chain, test_gain_int16_8_stereo_16000);
+  tcase_add_test (tc_chain, test_gain_int16_8_stereo_22050);
+  tcase_add_test (tc_chain, test_gain_int16_8_stereo_24000);
+  tcase_add_test (tc_chain, test_gain_int16_8_stereo_32000);
+  tcase_add_test (tc_chain, test_gain_int16_8_stereo_44100);
+  tcase_add_test (tc_chain, test_gain_int16_8_stereo_48000);
+
+  return s;
+}
+
+int
+main (int argc, char **argv)
+{
+  gint nf;
+
+  Suite *s = rganalysis_suite ();
+  SRunner *sr = srunner_create (s);
+
+  gst_check_init (&argc, &argv);
+
+  srunner_run_all (sr, CK_ENV);
+  nf = srunner_ntests_failed (sr);
+  srunner_free (sr);
+
+  return nf;
+}
diff -Naur gst-plugins-bad-orig/tests/check/Makefile.am gst-plugins-bad/tests/check/Makefile.am
--- gst-plugins-bad-orig/tests/check/Makefile.am	2006-09-15 08:19:37.000000000 +0200
+++ gst-plugins-bad/tests/check/Makefile.am	2006-09-30 15:33:37.000000000 +0200
@@ -52,6 +52,7 @@
 
 check_PROGRAMS = \
 	$(check_mpeg2enc)  \
+	elements/rganalysis \
 	elements/videocrop \
 	$(check_wavpack)
 
