Add replaygain playback elements (#412710).

Original commit message from CVS: Patch by: René Stadler <mail at renestadler de> * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/inspect/plugin-replaygain.xml: * gst/replaygain/Makefile.am: * gst/replaygain/gstrganalysis.c: (gst_rg_analysis_class_init), (gst_rg_analysis_start), (gst_rg_analysis_set_caps), (gst_rg_analysis_transform_ip), (gst_rg_analysis_event), (gst_rg_analysis_stop), (gst_rg_analysis_handle_tags), (gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result), (gst_rg_analysis_album_result): * gst/replaygain/gstrganalysis.h: * gst/replaygain/gstrglimiter.c: (gst_rg_limiter_base_init), (gst_rg_limiter_class_init), (gst_rg_limiter_init), (gst_rg_limiter_set_property), (gst_rg_limiter_get_property), (gst_rg_limiter_transform_ip): * gst/replaygain/gstrglimiter.h: * gst/replaygain/gstrgvolume.c: (gst_rg_volume_base_init), (gst_rg_volume_class_init), (gst_rg_volume_init), (gst_rg_volume_set_property), (gst_rg_volume_get_property), (gst_rg_volume_dispose), (gst_rg_volume_change_state), (gst_rg_volume_sink_event), (gst_rg_volume_tag_event), (gst_rg_volume_reset), (gst_rg_volume_update_gain), (gst_rg_volume_determine_gain): * gst/replaygain/gstrgvolume.h: * gst/replaygain/replaygain.c: (plugin_init): * gst/replaygain/replaygain.h: * gst/replaygain/rganalysis.h: * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/rganalysis.c: (send_eos_event), (GST_START_TEST): * tests/check/elements/rglimiter.c: (setup_rglimiter), (cleanup_rglimiter), (set_playing_state), (create_test_buffer), (verify_test_buffer), (GST_START_TEST), (rglimiter_suite), (main): * tests/check/elements/rgvolume.c: (event_func), (setup_rgvolume), (cleanup_rgvolume), (set_playing_state), (set_null_state), (send_eos_event), (send_tag_event), (test_buffer_new), (fail_unless_target_gain), (fail_unless_result_gain), (fail_unless_gain), (GST_START_TEST), (rgvolume_suite), (main): Add replaygain playback elements (#412710).
author: René Stadler <mail@renestadler.de> 2007-05-19 10:01:45 +0000
committer: Tim-Philipp Müller <tim@centricular.net> 2007-05-19 10:01:45 +0000
commit: 4e45e0a2690605843115abbbde06ca00a8da8584 (patch)
tree: addb448253cd26656ecf7bba66e87f73f6ff0f61
parent: fc99abef7f1ca971b919b433e87ed46cbd644aae (diff)
13 files changed, 2212 insertions, 240 deletions
diff --git a/gst/replaygain/Makefile.am b/gst/replaygain/Makefile.am
index d4523654..a0a3ca5a 100644
--- a/gst/replaygain/Makefile.am
+++ b/gst/replaygain/Makefile.am
@@ -2,12 +2,20 @@ plugin_LTLIBRARIES = libgstreplaygain.la
 
 libgstreplaygain_la_SOURCES = \
 	gstrganalysis.c       \
+	gstrglimiter.c        \
+	gstrgvolume.c         \
+	replaygain.c          \
 	rganalysis.c
-libgstreplaygain_la_CFLAGS  = $(GST_CFLAGS) $(GST_BASE_CFLAGS)
-libgstreplaygain_la_LIBADD  = $(GST_LIBS) $(GST_BASE_LIBS) $(LIBM)
+libgstreplaygain_la_CFLAGS  = \
+	$(GST_CFLAGS) $(GST_BASE_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
+libgstreplaygain_la_LIBADD  = \
+	$(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstpbutils-0.10 $(LIBM)
 libgstreplaygain_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
 
 # headers we need but don't want installed
 noinst_HEADERS =         \
 	gstrganalysis.h  \
+	gstrglimiter.h   \
+	gstrgvolume.h    \
+	replaygain.h     \
 	rganalysis.h
diff --git a/gst/replaygain/gstrganalysis.c b/gst/replaygain/gstrganalysis.c
index 9ad50e0d..24367786 100644
--- a/gst/replaygain/gstrganalysis.c
+++ b/gst/replaygain/gstrganalysis.c
@@ -22,103 +22,29 @@
 
 /**
  * SECTION:element-rganalysis
+ * @see_also: <link linkend="GstRgVolume">rgvolume</link>
  *
  * <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.
+ * This element analyzes raw audio sample data in accordance with the proposed
+ * <ulink url="http://replaygain.org">ReplayGain standard</ulink> for
+ * calculating the ideal replay gain for music tracks and albums.  The element
+ * is designed as a pass-through filter that never modifies any data.  As it
+ * receives an EOS event, it finalizes the ongoing analysis and generates a tag
+ * list containing the results.  It is sent downstream with a tag event and
+ * posted on the message bus with a tag message.  The EOS event is forwarded as
+ * normal afterwards.  Result tag lists at least contain the tags
+ * #GST_TAG_TRACK_GAIN, #GST_TAG_TRACK_PEAK and #GST_TAG_REFERENCE_LEVEL.
  * </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.
+ * 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 <link linkend="GstRgAnalysis--num-tracks">album processing</link>
+ * since the album gain and peak values need to be associated with all tracks of
+ * an album, not just the last one.
  * </para>
  * <title>Example launch lines</title>
  * <para>Analyze a simple test waveform:</para>
@@ -127,18 +53,26 @@
  * </programlisting>
  * <para>Analyze a given file:</para>
  * <programlisting>
- * gst-launch -t filesrc location="Some file.ogg" ! decodebin ! audioconvert ! audioresample ! rganalysis ! fakesink
+ * 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
+ * gst-launch -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav \
+ *     ! wavparse ! rganalysis ! fakesink
  * </programlisting>
+ * <para>
+ * The above launch line yields a result gain of +6 dB (instead of the expected
+ * +0 dB).  This is not in error, refer to the <link
+ * linkend="GstRgAnalysis--reference-level">reference-level</link> property
+ * documentation for more information.
+ * </para>
  * <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.
+ * 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>
  */
@@ -147,11 +81,11 @@
 #include <config.h>
 #endif
 
-#include <string.h>
 #include <gst/gst.h>
 #include <gst/base/gstbasetransform.h>
 
 #include "gstrganalysis.h"
+#include "replaygain.h"
 
 GST_DEBUG_CATEGORY_STATIC (gst_rg_analysis_debug);
 #define GST_CAT_DEFAULT gst_rg_analysis_debug
@@ -254,18 +188,93 @@ gst_rg_analysis_class_init (GstRgAnalysisClass * klass)
   gobject_class->set_property = gst_rg_analysis_set_property;
   gobject_class->get_property = gst_rg_analysis_get_property;
 
+  /**
+   * GstRgAnalysis:num-tracks:
+   *
+   * Number of remaining album tracks.
+   * 
+   * Analyzing several streams sequentially and assigning them a common result
+   * gain is known as "album processing".  If this gain is used during playback
+   * (by switching to "album mode"), all tracks of an album receive the same
+   * amplification.  This keeps the relative volume levels between the tracks
+   * intact.  To enable this, set this property to the number of streams that
+   * will be processed as album tracks.
+   *
+   * Every time an EOS event is received, the value of this property is
+   * decremented by one.  As it reaches zero, it is assumed that the last track
+   * of the album finished.  The tag list for the final stream will contain the
+   * additional tags #GST_TAG_ALBUM_GAIN and #GST_TAG_ALBUM_PEAK.  All other
+   * streams just get the two track tags posted because the values for the album
+   * tags are not known before all tracks are analyzed.  Applications need to
+   * ensure that the album gain and peak values are also associated with the
+   * other tracks when storing the results.
+   *
+   * If the total number of album tracks is unknown beforehand, just ensure that
+   * the value is greater than 1 before each track starts.  Then before the end
+   * of the last track, set it to the value 1.
+   *
+   * To perform album processing, the element has to preserve data between
+   * streams.  This cannot survive a state change to the NULL or READY state.
+   * If you change your pipeline's state to NULL or READY between tracks, lock
+   * the element's state using gst_element_set_locked_state() when it is in
+   * PAUSED or PLAYING.
+   */
   g_object_class_install_property (gobject_class, PROP_NUM_TRACKS,
       g_param_spec_int ("num-tracks", "Number of album tracks",
-          "Number of remaining tracks in the album",
-          0, G_MAXINT, 0, G_PARAM_READWRITE));
+          "Number of remaining album tracks", 0, G_MAXINT, 0,
+          G_PARAM_READWRITE));
+  /**
+   * GstRgAnalysis:forced:
+   *
+   * Whether to analyze streams even when ReplayGain tags exist.
+   *
+   * For assisting transcoder/converter applications, the element can silently
+   * skip the processing of streams that already contain the necessary tags.
+   * Data will flow as usual but the element will not consume CPU time and will
+   * not generate result tags.  To enable possible skipping, set this property
+   * to #FALSE.
+   *
+   * If used in conjunction with <link linkend="GstRgAnalysis--num-tracks">album
+   * processing</link>, the element will skip the number of remaining album
+   * tracks if a full set of tags is found for the first track.  If a subsequent
+   * track of the album is missing tags, processing cannot start again.  If this
+   * is undesired, the application has to scan all files beforehand and enable
+   * forcing of processing if needed.
+   */
   g_object_class_install_property (gobject_class, PROP_FORCED,
-      g_param_spec_boolean ("forced", "Force processing",
-          "Analyze streams even when ReplayGain tags exist",
+      g_param_spec_boolean ("forced", "Forced",
+          "Analyze even if ReplayGain tags exist",
           FORCED_DEFAULT, G_PARAM_READWRITE));
+  /**
+   * GstRgAnalysis:reference-level:
+   *
+   * Reference level [dB].
+   *
+   * Analyzing the ReplayGain pink noise reference waveform computes a result of
+   * +6 dB instead of the expected 0 dB.  This is because the default reference
+   * level is 89 dB.  To obtain values as lined out in the original proposal of
+   * ReplayGain, set this property to 83.
+   *
+   * Almost all software uses 89 dB as a reference however, and this value has
+   * become the new official value.  That is to say, while the change has been
+   * acclaimed by the author of the ReplayGain proposal, the <ulink
+   * url="http://replaygain.org">webpage</ulink> is still outdated at the time
+   * of this writing.
+   *
+   * The value was changed because the original proposal recommends a default
+   * pre-amp value of +6 dB for playback.  This seemed a bit odd, as it means
+   * that the algorithm has the general tendency to produce adjustment values
+   * that are 6 dB too low.  Bumping the reference level by 6 dB compensated for
+   * this.
+   *
+   * The problem of the reference level being ambiguous for lack of concise
+   * standardization is to be solved by adopting the #GST_TAG_REFERENCE_LEVEL
+   * tag, which allows to store the used value alongside the gain values.
+   */
   g_object_class_install_property (gobject_class, PROP_REFERENCE_LEVEL,
       g_param_spec_double ("reference-level", "Reference level",
-          "Reference level in dB (83.0 for original proposal)",
-          0.0, G_MAXDOUBLE, RG_REFERENCE_LEVEL, G_PARAM_READWRITE));
+          "Reference level [dB]", 0.0, 150., RG_REFERENCE_LEVEL,
+          G_PARAM_READWRITE));
 
   trans_class = (GstBaseTransformClass *) klass;
   trans_class->start = GST_DEBUG_FUNCPTR (gst_rg_analysis_start);
@@ -346,7 +355,7 @@ gst_rg_analysis_start (GstBaseTransform * base)
   filter->ctx = rg_analysis_new ();
   filter->analyze = NULL;
 
-  GST_DEBUG_OBJECT (filter, "Started");
+  GST_LOG_OBJECT (filter, "started");
 
   return TRUE;
 }
@@ -357,7 +366,7 @@ gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
 {
   GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
   GstStructure *structure;
-  const gchar *mime_type;
+  const gchar *name;
   gint n_channels, sample_rate, sample_bit_size, sample_size;
 
   g_return_val_if_fail (filter->ctx != NULL, FALSE);
@@ -367,7 +376,7 @@ gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
       in_caps, out_caps);
 
   structure = gst_caps_get_structure (in_caps, 0);
-  mime_type = gst_structure_get_name (structure);
+  name = gst_structure_get_name (structure);
 
   if (!gst_structure_get_int (structure, "width", &sample_bit_size)
       || !gst_structure_get_int (structure, "channels", &n_channels)
@@ -381,7 +390,7 @@ gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
     goto invalid_format;
   sample_size = sample_bit_size / 8;
 
-  if (strcmp (mime_type, "audio/x-raw-float") == 0) {
+  if (g_str_equal (name, "audio/x-raw-float")) {
 
     if (sample_size != sizeof (gfloat))
       goto invalid_format;
@@ -398,7 +407,7 @@ gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
     else
       goto invalid_format;
 
-  } else if (strcmp (mime_type, "audio/x-raw-int") == 0) {
+  } else if (g_str_equal (name, "audio/x-raw-int")) {
 
     if (sample_size != sizeof (gint16))
       goto invalid_format;
@@ -437,13 +446,13 @@ 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);
+  g_return_val_if_fail (filter->ctx != NULL, GST_FLOW_WRONG_STATE);
+  g_return_val_if_fail (filter->analyze != NULL, GST_FLOW_NOT_NEGOTIATED);
 
   if (filter->skip)
     return GST_FLOW_OK;
 
-  GST_DEBUG_OBJECT (filter, "Processing buffer of size %u",
+  GST_LOG_OBJECT (filter, "processing buffer of size %u",
       GST_BUFFER_SIZE (buf));
 
   filter->analyze (filter->ctx, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
@@ -463,11 +472,11 @@ gst_rg_analysis_event (GstBaseTransform * base, GstEvent * event)
 
     case GST_EVENT_EOS:
     {
-      GST_DEBUG_OBJECT (filter, "Received EOS event");
+      GST_LOG_OBJECT (filter, "received EOS event");
 
       gst_rg_analysis_handle_eos (filter);
 
-      GST_DEBUG_OBJECT (filter, "Passing on EOS event");
+      GST_LOG_OBJECT (filter, "passing on EOS event");
 
       break;
     }
@@ -498,7 +507,7 @@ gst_rg_analysis_stop (GstBaseTransform * base)
   rg_analysis_destroy (filter->ctx);
   filter->ctx = NULL;
 
-  GST_DEBUG_OBJECT (filter, "Stopped");
+  GST_LOG_OBJECT (filter, "stopped");
 
   return TRUE;
 }
@@ -514,13 +523,13 @@ gst_rg_analysis_handle_tags (GstRgAnalysis * filter,
     filter->ignore_tags = FALSE;
 
   if (filter->skip && album_processing) {
-    GST_INFO_OBJECT (filter, "Ignoring TAG event: Skipping album");
+    GST_DEBUG_OBJECT (filter, "ignoring tag event: skipping album");
     return;
   } else if (filter->skip) {
-    GST_INFO_OBJECT (filter, "Ignoring TAG event: Skipping track");
+    GST_DEBUG_OBJECT (filter, "ignoring tag event: skipping track");
     return;
   } else if (filter->ignore_tags) {
-    GST_INFO_OBJECT (filter, "Ignoring TAG event: Cannot skip anyways");
+    GST_DEBUG_OBJECT (filter, "ignoring tag event: cannot skip anyways");
     return;
   }
 
@@ -534,30 +543,31 @@ gst_rg_analysis_handle_tags (GstRgAnalysis * filter,
       GST_TAG_ALBUM_PEAK, &dummy);
 
   if (!(filter->has_track_gain && filter->has_track_peak)) {
-    GST_INFO_OBJECT (filter, "Track tags not complete yet");
+    GST_DEBUG_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");
+    GST_DEBUG_OBJECT (filter, "album tags not complete yet");
     return;
   }
 
   if (filter->forced) {
-    GST_INFO_OBJECT (filter,
-        "Existing tags are sufficient, but processing anyway (forced)");
+    GST_DEBUG_OBJECT (filter,
+        "existing tags are sufficient, but processing anyway (forced)");
     return;
   }
 
   filter->skip = TRUE;
   rg_analysis_reset (filter->ctx);
 
-  if (!album_processing)
-    GST_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");
+  if (!album_processing) {
+    GST_DEBUG_OBJECT (filter,
+        "existing tags are sufficient, will not process this track");
+  } else {
+    GST_DEBUG_OBJECT (filter,
+        "existing tags are sufficient, will not process this album");
+  }
 }
 
 static void
@@ -599,7 +609,9 @@ gst_rg_analysis_handle_eos (GstRgAnalysis * filter)
       rg_analysis_reset_album (filter->ctx);
 
     if (track_success || album_success) {
-      GST_DEBUG_OBJECT (filter, "Posting tag list with results");
+      GST_LOG_OBJECT (filter, "posting tag list with results");
+      gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
+          GST_TAG_REFERENCE_LEVEL, filter->reference_level, NULL);
       /* This 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);
@@ -609,11 +621,12 @@ gst_rg_analysis_handle_eos (GstRgAnalysis * filter)
   if (album_processing) {
     filter->num_tracks--;
 
-    if (!album_finished)
-      GST_INFO_OBJECT (filter, "Album not finished yet (num-tracks is now %u)",
+    if (!album_finished) {
+      GST_DEBUG_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)");
+    } else {
+      GST_DEBUG_OBJECT (filter, "album finished (num-tracks is now 0)");
+    }
   }
 
   if (album_processing)
@@ -631,10 +644,10 @@ gst_rg_analysis_track_result (GstRgAnalysis * filter, GstTagList ** tag_list)
 
   if (track_success) {
     track_gain += filter->reference_level - RG_REFERENCE_LEVEL;
-    GST_INFO_OBJECT (filter, "Track gain is %+.2f dB, peak %.6f", track_gain,
+    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");
+    GST_INFO_OBJECT (filter, "track was too short to analyze");
   }
 
   if (track_success) {
@@ -658,10 +671,10 @@ gst_rg_analysis_album_result (GstRgAnalysis * filter, GstTagList ** tag_list)
 
   if (album_success) {
     album_gain += filter->reference_level - RG_REFERENCE_LEVEL;
-    GST_INFO_OBJECT (filter, "Album gain is %+.2f dB, peak %.6f", album_gain,
+    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");
+    GST_INFO_OBJECT (filter, "album was too short to analyze");
   }
 
   if (album_success) {
@@ -673,14 +686,3 @@ gst_rg_analysis_album_result (GstRgAnalysis * filter, GstTagList ** tag_list)
 
   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, "replaygain",
-    "ReplayGain analysis", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME,
-    GST_PACKAGE_ORIGIN);
diff --git a/gst/replaygain/gstrganalysis.h b/gst/replaygain/gstrganalysis.h
index 121ce4af..fbf46830 100644
--- a/gst/replaygain/gstrganalysis.h
+++ b/gst/replaygain/gstrganalysis.h
@@ -78,6 +78,8 @@ struct _GstRgAnalysisClass
   GstBaseTransformClass parent_class;
 };
 
+GType gst_rg_analysis_get_type (void);
+
 G_END_DECLS
 
 #endif /* __GST_RG_ANALYSIS_H__ */
diff --git a/gst/replaygain/gstrglimiter.c b/gst/replaygain/gstrglimiter.c
new file mode 100644
index 00000000..609db3d7
--- /dev/null
+++ b/gst/replaygain/gstrglimiter.c
@@ -0,0 +1,197 @@
+/* GStreamer ReplayGain limiter
+ *
+ * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
+ * 
+ * gstrglimiter.c: Element to apply signal compression to raw audio data
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 of
+ * the License, or (at your option) any later version.
+ * 
+ * This library is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+/**
+ * SECTION:element-rglimiter
+ * @see_also: <link linkend="GstRgVolume">rgvolume</link>
+ *
+ * <refsect2>
+ * <para>
+ * This element applies signal compression/limiting to raw audio data.  It
+ * performs strict hard limiting with soft-knee characteristics, using a
+ * threshold of -6 dB.  This type of filter is mentioned in the proposed <ulink
+ * url="http://replaygain.org">ReplayGain standard</ulink>.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>Playback of a file:</para>
+ * <programlisting>
+ * gst-launch filesrc location="Filename.ext" ! decodebin ! audioconvert \
+ *            ! rgvolume pre-amp=6.0 headroom=10.0 ! rglimiter \
+ *            ! audioconvert ! audioresample ! alsasink
+ * </programlisting>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+#include <math.h>
+
+#include "gstrglimiter.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_rg_limiter_debug);
+#define GST_CAT_DEFAULT gst_rg_limiter_debug
+
+enum
+{
+  PROP_0,
+  PROP_ENABLED,
+};
+
+static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
+        "width = (int) 32, channels = (int) [1, MAX], "
+        "rate = (int) [1, MAX], endianness = (int) BYTE_ORDER"));
+
+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, channels = (int) [1, MAX], "
+        "rate = (int) [1, MAX], endianness = (int) BYTE_ORDER"));
+
+GST_BOILERPLATE (GstRgLimiter, gst_rg_limiter, GstBaseTransform,
+    GST_TYPE_BASE_TRANSFORM);
+
+static void gst_rg_limiter_class_init (GstRgLimiterClass * klass);
+static void gst_rg_limiter_init (GstRgLimiter * filter,
+    GstRgLimiterClass * gclass);
+
+static void gst_rg_limiter_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rg_limiter_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+
+static GstFlowReturn gst_rg_limiter_transform_ip (GstBaseTransform * base,
+    GstBuffer * buf);
+
+static const GstElementDetails element_details = {
+  "ReplayGain limiter",
+  "Filter/Effect/Audio",
+  "Apply signal compression to raw audio data",
+  "Ren\xc3\xa9 Stadler <mail@renestadler.de>"
+};
+
+static void
+gst_rg_limiter_base_init (gpointer g_class)
+{
+  GstElementClass *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, &element_details);
+
+  GST_DEBUG_CATEGORY_INIT (gst_rg_limiter_debug, "rglimiter", 0,
+      "ReplayGain limiter element");
+}
+
+static void
+gst_rg_limiter_class_init (GstRgLimiterClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstBaseTransformClass *trans_class;
+
+  gobject_class = (GObjectClass *) klass;
+
+  gobject_class->set_property = gst_rg_limiter_set_property;
+  gobject_class->get_property = gst_rg_limiter_get_property;
+
+  g_object_class_install_property (gobject_class, PROP_ENABLED,
+      g_param_spec_boolean ("enabled", "Enabled", "Enable processing", TRUE,
+          G_PARAM_READWRITE));
+
+  trans_class = GST_BASE_TRANSFORM_CLASS (klass);
+  trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_limiter_transform_ip);
+  trans_class->passthrough_on_same_caps = FALSE;
+}
+
+static void
+gst_rg_limiter_init (GstRgLimiter * filter, GstRgLimiterClass * gclass)
+{
+  filter->enabled = TRUE;
+  gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter), FALSE);
+}
+
+static void
+gst_rg_limiter_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRgLimiter *filter = GST_RG_LIMITER (object);
+
+  switch (prop_id) {
+    case PROP_ENABLED:
+      filter->enabled = g_value_get_boolean (value);
+      gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
+          !filter->enabled);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rg_limiter_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRgLimiter *filter = GST_RG_LIMITER (object);
+
+  switch (prop_id) {
+    case PROP_ENABLED:
+      g_value_set_boolean (value, filter->enabled);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+#define LIMIT 1.0
+#define THRES 0.5               /* ca. -6 dB */
+#define COMPL 0.5               /* LIMIT - THRESH */
+
+static GstFlowReturn
+gst_rg_limiter_transform_ip (GstBaseTransform * base, GstBuffer * buf)
+{
+  GstRgLimiter *filter = GST_RG_LIMITER (base);
+  gfloat *input;
+  guint count;
+  guint i;
+
+  if (!filter->enabled)
+    return GST_FLOW_OK;
+
+  input = (gfloat *) GST_BUFFER_DATA (buf);
+  count = GST_BUFFER_SIZE (buf) / sizeof (gfloat);
+
+  for (i = count; i--;) {
+    if (*input > THRES)
+      *input = tanhf ((*input - THRES) / COMPL) * COMPL + THRES;
+    else if (*input < -THRES)
+      *input = tanhf ((*input + THRES) / COMPL) * COMPL - THRES;
+    input++;
+  }
+
+  return GST_FLOW_OK;
+}
diff --git a/gst/replaygain/gstrglimiter.h b/gst/replaygain/gstrglimiter.h
new file mode 100644
index 00000000..63bd8049
--- /dev/null
+++ b/gst/replaygain/gstrglimiter.h
@@ -0,0 +1,64 @@
+/* GStreamer ReplayGain limiter
+ *
+ * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
+ *
+ * gstrglimiter.h: Element to apply signal compression to raw audio data
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 of
+ * the License, or (at your option) any later version.
+ * 
+ * This library is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifndef __GST_RG_LIMITER_H__
+#define __GST_RG_LIMITER_H__
+
+#include <gst/gst.h>
+#include <gst/base/gstbasetransform.h>
+
+#define GST_TYPE_RG_LIMITER \
+  (gst_rg_limiter_get_type())
+#define GST_RG_LIMITER(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_LIMITER,GstRgLimiter))
+#define GST_RG_LIMITER_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_LIMITER,GstRgLimiterClass))
+#define GST_IS_RG_LIMITER(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_LIMITER))
+#define GST_IS_RG_LIMITER_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_LIMITER))
+
+typedef struct _GstRgLimiter GstRgLimiter;
+typedef struct _GstRgLimiterClass GstRgLimiterClass;
+
+/**
+ * GstRgLimiter:
+ *
+ * Opaque data structure.
+ */
+struct _GstRgLimiter
+{
+  GstBaseTransform element;
+
+  /*< private >*/
+
+  gboolean enabled;
+};
+
+struct _GstRgLimiterClass
+{
+  GstBaseTransformClass parent_class;
+};
+
+GType gst_rg_limiter_get_type (void);
+
+#endif /* __GST_RG_LIMITER_H__ */
diff --git a/gst/replaygain/gstrgvolume.c b/gst/replaygain/gstrgvolume.c
new file mode 100644
index 00000000..35b4f5ef
--- /dev/null
+++ b/gst/replaygain/gstrgvolume.c
@@ -0,0 +1,702 @@
+/* GStreamer ReplayGain volume adjustment
+ *
+ * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
+ * 
+ * gstrgvolume.c: Element to apply ReplayGain volume adjustment
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 of
+ * the License, or (at your option) any later version.
+ * 
+ * This library is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+/**
+ * SECTION:element-rgvolume
+ * @see_also: <link linkend="GstRgLimiter">rglimiter</link>,
+ *            <link linkend="GstRgAnalysis">rganalysis</link>
+ *
+ * <refsect2>
+ * <para>
+ * This element applies volume changes to streams as lined out in the proposed
+ * <ulink url="http://replaygain.org">ReplayGain standard</ulink>.  It
+ * interprets the ReplayGain meta data tags and carries out the adjustment (by
+ * using a volume element internally).  The relevant tags are:
+ * <itemizedlist>
+ * <listitem>#GST_TAG_TRACK_GAIN</listitem>
+ * <listitem>#GST_TAG_TRACK_PEAK</listitem>
+ * <listitem>#GST_TAG_ALBUM_GAIN</listitem>
+ * <listitem>#GST_TAG_ALBUM_PEAK</listitem>
+ * <listitem>#GST_TAG_REFERENCE_LEVEL</listitem>
+ * </itemizedlist>
+ * The information carried by these tags must have been calculated beforehand by
+ * performing the ReplayGain analysis.  This is implemented by the <link
+ * linkend="GstRgAnalysis">rganalysis</link> element.
+ * </para>
+ * <para>
+ * The signal compression/limiting recommendations outlined in the proposed
+ * standard are not implemented by this element.  This has to be handled by
+ * separate elements because applications might want to have additional filters
+ * between the volume adjustment and the limiting stage.  A basic limiter is
+ * included with this plugin: The <link linkend="GstRgLimiter">rglimiter</link>
+ * element applies -6 dB hard limiting as mentioned in the ReplayGain standard.
+ * </para>
+ * <title>Example launch line</title>
+ * <para>Playback of a file:</para>
+ * <programlisting>
+ * gst-launch filesrc location="Filename.ext" ! decodebin ! audioconvert \
+ *     ! rgvolume ! audioconvert ! audioresample ! alsasink
+ * </programlisting>
+ * </refsect2>
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+#include <gst/pbutils/pbutils.h>
+#include <math.h>
+
+#include "gstrgvolume.h"
+#include "replaygain.h"
+
+GST_DEBUG_CATEGORY_STATIC (gst_rg_volume_debug);
+#define GST_CAT_DEFAULT gst_rg_volume_debug
+
+enum
+{
+  PROP_0,
+  PROP_ALBUM_MODE,
+  PROP_HEADROOM,
+  PROP_PRE_AMP,
+  PROP_FALLBACK_GAIN,
+  PROP_TARGET_GAIN,
+  PROP_RESULT_GAIN
+};
+
+#define DEFAULT_ALBUM_MODE TRUE
+#define DEFAULT_HEADROOM 0.0
+#define DEFAULT_PRE_AMP 0.0
+#define DEFAULT_FALLBACK_GAIN 0.0
+
+#define DB_TO_LINEAR(x) pow (10., (x) / 20.)
+#define LINEAR_TO_DB(x) (20. * log10 (x))
+
+#define GAIN_FORMAT "+.02f dB"
+#define PEAK_FORMAT ".06f"
+
+#define VALID_GAIN(x) ((x) > -60.00 && (x) < 60.00)
+#define VALID_PEAK(x) ((x) > 0.)
+
+/* Same template caps as GstVolume, for I don't like having just ANY caps. */
+
+static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
+        "rate = (int) [ 1, MAX ], "
+        "channels = (int) [ 1, MAX ], "
+        "endianness = (int) BYTE_ORDER, "
+        "width = (int) 32; "
+        "audio/x-raw-int, "
+        "channels = (int) [ 1, MAX ], "
+        "rate = (int) [ 1,  MAX ], "
+        "endianness = (int) BYTE_ORDER, "
+        "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE"));
+
+static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
+        "rate = (int) [ 1, MAX ], "
+        "channels = (int) [ 1, MAX ], "
+        "endianness = (int) BYTE_ORDER, "
+        "width = (int) 32; "
+        "audio/x-raw-int, "
+        "channels = (int) [ 1, MAX ], "
+        "rate = (int) [ 1,  MAX ], "
+        "endianness = (int) BYTE_ORDER, "
+        "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE"));
+
+GST_BOILERPLATE (GstRgVolume, gst_rg_volume, GstBin, GST_TYPE_BIN);
+
+static void gst_rg_volume_class_init (GstRgVolumeClass * klass);
+static void gst_rg_volume_init (GstRgVolume * self, GstRgVolumeClass * gclass);
+
+static void gst_rg_volume_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec);
+static void gst_rg_volume_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec);
+static void gst_rg_volume_dispose (GObject * object);
+
+static GstStateChangeReturn gst_rg_volume_change_state (GstElement * element,
+    GstStateChange transition);
+static gboolean gst_rg_volume_sink_event (GstPad * pad, GstEvent * event);
+
+static GstEvent *gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event);
+static void gst_rg_volume_reset (GstRgVolume * self);
+static void gst_rg_volume_update_gain (GstRgVolume * self);
+static inline void gst_rg_volume_determine_gain (GstRgVolume * self,
+    gdouble * target_gain, gdouble * result_gain);
+
+static void
+gst_rg_volume_base_init (gpointer g_class)
+{
+  GstElementClass *element_class = g_class;
+
+  static const GstElementDetails element_details = {
+    "ReplayGain volume",
+    "Filter/Effect/Audio",
+    "Apply ReplayGain volume adjustment",
+    "Ren\xc3\xa9 Stadler <mail@renestadler.de>"
+  };
+
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&src_template));
+  gst_element_class_add_pad_template (element_class,
+      gst_static_pad_template_get (&sink_template));
+  gst_element_class_set_details (element_class, &element_details);
+
+  GST_DEBUG_CATEGORY_INIT (gst_rg_volume_debug, "rgvolume", 0,
+      "ReplayGain volume element");
+}
+
+static void
+gst_rg_volume_class_init (GstRgVolumeClass * klass)
+{
+  GObjectClass *gobject_class;
+  GstElementClass *element_class;
+  GstBinClass *bin_class;
+
+  gobject_class = (GObjectClass *) klass;
+
+  gobject_class->set_property = gst_rg_volume_set_property;
+  gobject_class->get_property = gst_rg_volume_get_property;
+  gobject_class->dispose = gst_rg_volume_dispose;
+
+  /**
+   * GstRgVolume:album-mode:
+   *
+   * Whether to prefer album gain over track gain.
+   *
+   * If set to %TRUE, use album gain instead of track gain if both are
+   * available.  This keeps the relative loudness levels of tracks from the same
+   * album intact.
+   *
+   * If set to %FALSE, track mode is used instead.  This effectively leads to
+   * more extensive normalization.
+   *
+   * If album mode is enabled but the album gain tag is absent in the stream,
+   * the track gain is used instead.  If both gain tags are missing, the value
+   * of the <link linkend="GstRgVolume--fallback-gain">fallback-gain</link>
+   * property is used instead.
+   */
+  g_object_class_install_property (gobject_class, PROP_ALBUM_MODE,
+      g_param_spec_boolean ("album-mode", "Album mode",
+          "Prefer album over track gain", DEFAULT_ALBUM_MODE,
+          G_PARAM_READWRITE));
+  /**
+   * GstRgVolume:headroom:
+   *
+   * Extra headroom [dB].  This controls the amount by which the output can
+   * exceed digital full scale.
+   *
+   * Only set this to a value greater than 0.0 if signal compression/limiting of
+   * a suitable form is applied to the output (or output is brought into the
+   * correct range by some other transformation).
+   *
+   * This element internally uses a volume element, which also supports
+   * operating on integer audio formats.  These formats do not allow exceeding
+   * digital full scale.  If extra headroom is used, make sure that the raw
+   * audio data format is floating point (audio/x-raw-float).  Otherwise,
+   * clipping distortion might be introduced as part of the volume adjustment
+   * itself.
+   */
+  g_object_class_install_property (gobject_class, PROP_HEADROOM,
+      g_param_spec_double ("headroom", "Headroom", "Extra headroom [dB]",
+          0., 60., DEFAULT_HEADROOM, G_PARAM_READWRITE));
+  /**
+   * GstRgVolume:pre-amp:
+   *
+   * Additional gain to apply globally [dB].  This controls the trade-off
+   * between uniformity of normalization and utilization of available dynamic
+   * range.
+   *
+   * Note that the default value is 0 dB because the ReplayGain reference value
+   * was adjusted by +6 dB (from 83 to 89 dB).  At the time of this writing, the
+   * <ulink url="http://replaygain.org">webpage</ulink> is still outdated and
+   * does not reflect this change however.  Where the original proposal states
+   * that a proper default pre-amp value is +6 dB, this translates to the used 0
+   * dB.
+   */
+  g_object_class_install_property (gobject_class, PROP_PRE_AMP,
+      g_param_spec_double ("pre-amp", "Pre-amp", "Extra gain [dB]",
+          -60., 60., DEFAULT_PRE_AMP, G_PARAM_READWRITE));
+  /**
+   * GstRgVolume:fallback-gain:
+   *
+   * Fallback gain [dB] for streams missing ReplayGain tags.
+   */
+  g_object_class_install_property (gobject_class, PROP_FALLBACK_GAIN,
+      g_param_spec_double ("fallback-gain", "Fallback gain",
+          "Gain for streams missing tags [dB]",
+          -60., 60., DEFAULT_FALLBACK_GAIN, G_PARAM_READWRITE));
+  /**
+   * GstRgVolume:result-gain:
+   *
+   * Applied gain [dB].  This gain is applied to processed buffer data.
+   *
+   * This is set to the <link linkend="GstRgVolume--target-gain">target
+   * gain</link> if amplification by that amount can be applied safely.
+   * "Safely" means that the volume adjustment does not inflict clipping
+   * distortion.  Should this not be the case, the result gain is set to an
+   * appropriately reduced value (by applying peak normalization).  The proposed
+   * standard calls this "clipping prevention".
+   *
+   * The difference between target and result gain reflects the necessary amount
+   * of reduction.  Applications can make use of this information to temporarily
+   * reduce the <link linkend="GstRgVolume--pre-amp">pre-amp</link> for
+   * subsequent streams, as recommended by the ReplayGain standard.
+   *
+   * Note that target and result gain differing for a great majority of streams
+   * indicates a problem: What happens in this case is that most streams receive
+   * peak normalization instead of amplification by the ideal replay gain.  To
+   * prevent this, the <link linkend="GstRgVolume--pre-amp">pre-amp</link> has
+   * to be lowered and/or a limiter has to be used which facilitates the use of
+   * <link linkend="GstRgVolume--headroom">headroom</link>.
+   */
+  g_object_class_install_property (gobject_class, PROP_RESULT_GAIN,
+      g_param_spec_double ("result-gain", "Result-gain", "Applied gain [dB]",
+          -120., 120., 0., G_PARAM_READABLE));
+  /**
+   * GstRgVolume:target-gain:
+   *
+   * Applicable gain [dB].  This gain is supposed to be applied.
+   *
+   * Depending on the value of the <link
+   * linkend="GstRgVolume--album-mode">album-mode</link> property and the
+   * presence of ReplayGain tags in the stream, this is set according to one of
+   * these simple formulas:
+   *
+   * <itemizedlist>
+   * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + album gain
+   * of the stream</listitem>
+   * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + track gain
+   * of the stream</listitem>
+   * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + <link
+   * linkend="GstRgVolume--fallback-gain">fallback gain</link></listitem>
+   * </itemizedlist>
+   */
+  g_object_class_install_property (gobject_class, PROP_TARGET_GAIN,
+      g_param_spec_double ("target-gain", "Target-gain",
+          "Applicable gain [dB]", -120., 120., 0., G_PARAM_READABLE));
+
+  element_class = (GstElementClass *) klass;
+  element_class->change_state = GST_DEBUG_FUNCPTR (gst_rg_volume_change_state);
+
+  bin_class = (GstBinClass *) klass;
+  /* Setting these to NULL makes gst_bin_add and _remove refuse to let anyone
+   * mess with our internals. */
+  bin_class->add_element = NULL;
+  bin_class->remove_element = NULL;
+}
+
+static void
+gst_rg_volume_init (GstRgVolume * self, GstRgVolumeClass * gclass)
+{
+  GObjectClass *volume_class;
+  GstPad *volume_pad, *ghost_pad;
+
+  self->album_mode = DEFAULT_ALBUM_MODE;
+  self->headroom = DEFAULT_HEADROOM;
+  self->pre_amp = DEFAULT_PRE_AMP;
+  self->fallback_gain = DEFAULT_FALLBACK_GAIN;
+  self->target_gain = 0.0;
+  self->result_gain = 0.0;
+
+  self->volume_element = gst_element_factory_make ("volume", "rgvolume-volume");
+  if (G_UNLIKELY (self->volume_element == NULL)) {
+    GstMessage *msg;
+
+    GST_WARNING_OBJECT (self, "could not create volume element");
+    msg = gst_missing_element_message_new (GST_ELEMENT_CAST (self), "volume");
+    gst_element_post_message (GST_ELEMENT_CAST (self), msg);
+
+    /* Nothing else to do, we will refuse the state change from NULL to READY to
+     * indicate that something went very wrong.  It is doubtful that someone
+     * attempts changing our state though, since we end up having no pads! */
+    return;
+  }
+
+  volume_class = G_OBJECT_GET_CLASS (G_OBJECT (self->volume_element));
+  self->max_volume = G_PARAM_SPEC_DOUBLE
+      (g_object_class_find_property (volume_class, "volume"))->maximum;
+
+  GST_BIN_CLASS (parent_class)->add_element (GST_BIN_CAST (self),
+      self->volume_element);
+
+  volume_pad = gst_element_get_pad (self->volume_element, "sink");
+  ghost_pad = gst_ghost_pad_new_from_template ("sink", volume_pad,
+      gst_pad_get_pad_template (volume_pad));
+  gst_object_unref (volume_pad);
+  gst_pad_set_event_function (ghost_pad, gst_rg_volume_sink_event);
+  gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad);
+
+  volume_pad = gst_element_get_pad (self->volume_element, "src");
+  ghost_pad = gst_ghost_pad_new_from_template ("src", volume_pad,
+      gst_pad_get_pad_template (volume_pad));
+  gst_object_unref (volume_pad);
+  gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad);
+}
+
+static void
+gst_rg_volume_set_property (GObject * object, guint prop_id,
+    const GValue * value, GParamSpec * pspec)
+{
+  GstRgVolume *self = GST_RG_VOLUME (object);
+
+  switch (prop_id) {
+    case PROP_ALBUM_MODE:
+      self->album_mode = g_value_get_boolean (value);
+      break;
+    case PROP_HEADROOM:
+      self->headroom = g_value_get_double (value);
+      break;
+    case PROP_PRE_AMP:
+      self->pre_amp = g_value_get_double (value);
+      break;
+    case PROP_FALLBACK_GAIN:
+      self->fallback_gain = g_value_get_double (value);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+
+  gst_rg_volume_update_gain (self);
+}
+
+static void
+gst_rg_volume_get_property (GObject * object, guint prop_id,
+    GValue * value, GParamSpec * pspec)
+{
+  GstRgVolume *self = GST_RG_VOLUME (object);
+
+  switch (prop_id) {
+    case PROP_ALBUM_MODE:
+      g_value_set_boolean (value, self->album_mode);
+      break;
+    case PROP_HEADROOM:
+      g_value_set_double (value, self->headroom);
+      break;
+    case PROP_PRE_AMP:
+      g_value_set_double (value, self->pre_amp);
+      break;
+    case PROP_FALLBACK_GAIN:
+      g_value_set_double (value, self->fallback_gain);
+      break;
+    case PROP_TARGET_GAIN:
+      g_value_set_double (value, self->target_gain);
+      break;
+    case PROP_RESULT_GAIN:
+      g_value_set_double (value, self->result_gain);
+      break;
+    default:
+      G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
+      break;
+  }
+}
+
+static void
+gst_rg_volume_dispose (GObject * object)
+{
+  GstRgVolume *self = GST_RG_VOLUME (object);
+
+  if (self->volume_element != NULL) {
+    /* Manually remove our child using the bin implementation of remove_element.
+     * This is needed because we prevent gst_bin_remove from working, which the
+     * parent dispose handler would use if we had any children left. */
+    GST_BIN_CLASS (parent_class)->remove_element (GST_BIN_CAST (self),
+        self->volume_element);
+    self->volume_element = NULL;
+  }
+
+  G_OBJECT_CLASS (parent_class)->dispose (object);
+}
+
+static GstStateChangeReturn
+gst_rg_volume_change_state (GstElement * element, GstStateChange transition)
+{
+  GstRgVolume *self = GST_RG_VOLUME (element);
+  GstStateChangeReturn res;
+
+  switch (transition) {
+    case GST_STATE_CHANGE_NULL_TO_READY:
+
+      if (G_UNLIKELY (self->volume_element == NULL)) {
+        /* Creating our child volume element in _init failed. */
+        return GST_STATE_CHANGE_FAILURE;
+      }
+      break;
+
+    case GST_STATE_CHANGE_READY_TO_PAUSED:
+
+      gst_rg_volume_reset (self);
+      break;
+
+    default:
+      break;
+  }
+
+  res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
+
+  return res;
+}
+
+/* Event function for the ghost sink pad. */
+static gboolean
+gst_rg_volume_sink_event (GstPad * pad, GstEvent * event)
+{
+  GstRgVolume *self;
+  GstPad *volume_sink_pad;
+  GstEvent *send_event = event;
+  gboolean res;
+
+  self = GST_RG_VOLUME (gst_pad_get_parent_element (pad));
+  volume_sink_pad = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
+
+  switch (GST_EVENT_TYPE (event)) {
+    case GST_EVENT_TAG:
+
+      GST_LOG_OBJECT (self, "received tag event");
+
+      send_event = gst_rg_volume_tag_event (self, event);
+
+      if (send_event == NULL)
+        GST_LOG_OBJECT (self, "all tags handled, dropping event");
+
+      break;
+
+    case GST_EVENT_EOS:
+
+      gst_rg_volume_reset (self);
+      break;
+
+    default:
+      break;
+  }
+
+  if (G_LIKELY (send_event != NULL))
+    res = gst_pad_send_event (volume_sink_pad, send_event);
+  else
+    res = TRUE;
+
+  gst_object_unref (volume_sink_pad);
+  gst_object_unref (self);
+  return res;
+}
+
+static GstEvent *
+gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event)
+{
+  GstTagList *tag_list;
+  gboolean has_track_gain, has_track_peak, has_album_gain, has_album_peak;
+  gboolean has_ref_level;
+
+  g_return_val_if_fail (event != NULL, NULL);
+  g_return_val_if_fail (GST_EVENT_TYPE (event) == GST_EVENT_TAG, event);
+
+  gst_event_parse_tag (event, &tag_list);
+
+  if (gst_tag_list_is_empty (tag_list))
+    return event;
+
+  has_track_gain = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN,
+      &self->track_gain);
+  has_track_peak = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK,
+      &self->track_peak);
+  has_album_gain = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN,
+      &self->album_gain);
+  has_album_peak = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK,
+      &self->album_peak);
+  has_ref_level = gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
+      &self->reference_level);
+
+  if (!has_track_gain && !has_track_peak && !has_album_gain && !has_album_peak)
+    return event;
+
+  if (has_ref_level && (has_track_gain || has_album_gain)
+      && (ABS (self->reference_level - RG_REFERENCE_LEVEL) > 1.e-6)) {
+    /* Log a message stating the amount of adjustment that is applied below. */
+    GST_DEBUG_OBJECT (self,
+        "compensating for reference level difference by %" GAIN_FORMAT,
+        RG_REFERENCE_LEVEL - self->reference_level);
+  }
+  if (has_track_gain) {
+    self->track_gain += RG_REFERENCE_LEVEL - self->reference_level;
+  }
+  if (has_album_gain) {
+    self->album_gain += RG_REFERENCE_LEVEL - self->reference_level;
+  }
+
+  /* Ignore values that are obviously invalid. */
+  if (G_UNLIKELY (has_track_gain && !VALID_GAIN (self->track_gain))) {
+    GST_DEBUG_OBJECT (self,
+        "ignoring bogus track gain value %" GAIN_FORMAT, self->track_gain);
+    has_track_gain = FALSE;
+  }
+  if (G_UNLIKELY (has_track_peak && !VALID_PEAK (self->track_peak))) {
+    GST_DEBUG_OBJECT (self,
+        "ignoring bogus track peak value %" PEAK_FORMAT, self->track_peak);
+    has_track_peak = FALSE;
+  }
+  if (G_UNLIKELY (has_album_gain && !VALID_GAIN (self->album_gain))) {
+    GST_DEBUG_OBJECT (self,
+        "ignoring bogus album gain value %" GAIN_FORMAT, self->album_gain);
+    has_album_gain = FALSE;
+  }
+  if (G_UNLIKELY (has_album_peak && !VALID_PEAK (self->album_peak))) {
+    GST_DEBUG_OBJECT (self,
+        "ignoring bogus album peak value %" PEAK_FORMAT, self->album_peak);
+    has_album_peak = FALSE;
+  }
+
+  self->has_track_gain |= has_track_gain;
+  self->has_track_peak |= has_track_peak;
+  self->has_album_gain |= has_album_gain;
+  self->has_album_peak |= has_album_peak;
+
+  event = (GstEvent *) gst_mini_object_make_writable (GST_MINI_OBJECT (event));
+  gst_event_parse_tag (event, &tag_list);
+
+  gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_GAIN);
+  gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_PEAK);
+  gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_GAIN);
+  gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_PEAK);
+  gst_tag_list_remove_tag (tag_list, GST_TAG_REFERENCE_LEVEL);
+
+  gst_rg_volume_update_gain (self);
+
+  if (gst_tag_list_is_empty (tag_list)) {
+    gst_event_unref (event);
+    event = NULL;
+  }
+
+  return event;
+}
+
+static void
+gst_rg_volume_reset (GstRgVolume * self)
+{
+  self->has_track_gain = FALSE;
+  self->has_track_peak = FALSE;
+  self->has_album_gain = FALSE;
+  self->has_album_peak = FALSE;
+
+  self->reference_level = RG_REFERENCE_LEVEL;
+
+  gst_rg_volume_update_gain (self);
+}
+
+static void
+gst_rg_volume_update_gain (GstRgVolume * self)
+{
+  gdouble target_gain, result_gain, result_volume;
+  gboolean target_gain_changed, result_gain_changed;
+
+  gst_rg_volume_determine_gain (self, &target_gain, &result_gain);
+
+  result_volume = DB_TO_LINEAR (result_gain);
+
+  /* Ensure that the result volume is within the range that the volume element
+   * can handle.  Currently, the limit is 10. (+20 dB), which should not be
+   * restrictive. */
+  if (G_UNLIKELY (result_volume > self->max_volume)) {
+    GST_INFO_OBJECT (self,
+        "cannot handle result gain of %" GAIN_FORMAT " (%0.6f), adjusting",
+        result_gain, result_volume);
+
+    result_volume = self->max_volume;
+    result_gain = LINEAR_TO_DB (result_volume);
+  }
+
+  /* Direct comparison is OK in this case. */
+  if (target_gain == result_gain) {
+    GST_INFO_OBJECT (self,
+        "result gain is %" GAIN_FORMAT " (%0.6f), matching target",
+        result_gain, result_volume);
+  } else {
+    GST_INFO_OBJECT (self,
+        "result gain is %" GAIN_FORMAT " (%0.6f), target is %" GAIN_FORMAT,
+        result_gain, result_volume, target_gain);
+  }
+
+  target_gain_changed = (self->target_gain != target_gain);
+  result_gain_changed = (self->result_gain != result_gain);
+
+  self->target_gain = target_gain;
+  self->result_gain = result_gain;
+
+  g_object_set (self->volume_element, "volume", result_volume, NULL);
+
+  if (target_gain_changed)
+    g_object_notify ((GObject *) self, "target-gain");
+  if (result_gain_changed)
+    g_object_notify ((GObject *) self, "result-gain");
+}
+
+static inline void
+gst_rg_volume_determine_gain (GstRgVolume * self, gdouble * target_gain,
+    gdouble * result_gain)
+{
+  gdouble gain, peak;
+
+  if (!self->has_track_gain && !self->has_album_gain) {
+
+    GST_DEBUG_OBJECT (self, "using fallback gain");
+    gain = self->fallback_gain;
+    peak = 1.0;
+
+  } else if ((self->album_mode && self->has_album_gain)
+      || (!self->album_mode && !self->has_track_gain)) {
+
+    gain = self->album_gain;
+    if (G_LIKELY (self->has_album_peak)) {
+      peak = self->album_peak;
+    } else {
+      GST_DEBUG_OBJECT (self, "album peak missing, assuming 1.0");
+      peak = 1.0;
+    }
+    /* Falling back from track to album gain shouldn't really happen. */
+    if (G_UNLIKELY (!self->album_mode))
+      GST_INFO_OBJECT (self, "falling back to album gain");
+
+  } else {
+    /* !album_mode && !has_album_gain || album_mode && has_track_gain */
+
+    gain = self->track_gain;
+    if (G_LIKELY (self->has_track_peak)) {
+      peak = self->track_peak;
+    } else {
+      GST_DEBUG_OBJECT (self, "track peak missing, assuming 1.0");
+      peak = 1.0;
+    }
+    if (self->album_mode)
+      GST_INFO_OBJECT (self, "falling back to track gain");
+  }
+
+  gain += self->pre_amp;
+
+  *target_gain = gain;
+  *result_gain = gain;
+
+  if (LINEAR_TO_DB (peak) + gain > self->headroom) {
+    *result_gain = LINEAR_TO_DB (1. / peak) + self->headroom;
+  }
+}
diff --git a/gst/replaygain/gstrgvolume.h b/gst/replaygain/gstrgvolume.h
new file mode 100644
index 00000000..8fc29614
--- /dev/null
+++ b/gst/replaygain/gstrgvolume.h
@@ -0,0 +1,88 @@
+/* GStreamer ReplayGain volume adjustment
+ *
+ * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
+ *
+ * gstrgvolume.h: Element to apply ReplayGain volume adjustment
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 of
+ * the License, or (at your option) any later version.
+ * 
+ * This library is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifndef __GST_RG_VOLUME_H__
+#define __GST_RG_VOLUME_H__
+
+#include <gst/gst.h>
+
+G_BEGIN_DECLS
+
+#define GST_TYPE_RG_VOLUME \
+  (gst_rg_volume_get_type())
+#define GST_RG_VOLUME(obj) \
+  (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_VOLUME,GstRgVolume))
+#define GST_RG_VOLUME_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_VOLUME,GstRgVolumeClass))
+#define GST_IS_PLUGIN_TEMPLATE(obj) \
+  (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_VOLUME))
+#define GST_IS_PLUGIN_TEMPLATE_CLASS(klass) \
+  (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_VOLUME))
+
+typedef struct _GstRgVolume GstRgVolume;
+typedef struct _GstRgVolumeClass GstRgVolumeClass;
+
+/**
+ * GstRgVolume:
+ *
+ * Opaque data structure.
+ */
+struct _GstRgVolume
+{
+  GstBin bin;
+
+  /*< private >*/
+
+  GstElement *volume_element;
+  gdouble max_volume;
+
+  gboolean album_mode;
+  gdouble headroom;
+  gdouble pre_amp;
+  gdouble fallback_gain;
+
+  gdouble target_gain;
+  gdouble result_gain;
+
+  gdouble track_gain;
+  gdouble track_peak;
+  gdouble album_gain;
+  gdouble album_peak;
+
+  gboolean has_track_gain;
+  gboolean has_track_peak;
+  gboolean has_album_gain;
+  gboolean has_album_peak;
+
+  gdouble reference_level;
+};
+
+struct _GstRgVolumeClass
+{
+  GstBinClass parent_class;
+};
+
+GType gst_rg_volume_get_type (void);
+
+G_END_DECLS
+
+#endif /* __GST_RG_VOLUME_H__ */
diff --git a/gst/replaygain/replaygain.c b/gst/replaygain/replaygain.c
new file mode 100644
index 00000000..d0127e8b
--- /dev/null
+++ b/gst/replaygain/replaygain.c
@@ -0,0 +1,53 @@
+/* GStreamer ReplayGain plugin
+ *
+ * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
+ * 
+ * replaygain.c: Plugin providing ReplayGain related elements
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 of
+ * the License, or (at your option) any later version.
+ * 
+ * This library is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifdef HAVE_CONFIG_H
+#include <config.h>
+#endif
+
+#include <gst/gst.h>
+
+#include "gstrganalysis.h"
+#include "gstrglimiter.h"
+#include "gstrgvolume.h"
+
+static gboolean
+plugin_init (GstPlugin * plugin)
+{
+  if (!gst_element_register (plugin, "rganalysis", GST_RANK_NONE,
+          GST_TYPE_RG_ANALYSIS))
+    return FALSE;
+
+  if (!gst_element_register (plugin, "rglimiter", GST_RANK_NONE,
+          GST_TYPE_RG_LIMITER))
+    return FALSE;
+
+  if (!gst_element_register (plugin, "rgvolume", GST_RANK_NONE,
+          GST_TYPE_RG_VOLUME))
+    return FALSE;
+
+  return TRUE;
+}
+
+GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "replaygain",
+    "ReplayGain volume normalization", plugin_init, VERSION, GST_LICENSE,
+    GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
diff --git a/gst/replaygain/replaygain.h b/gst/replaygain/replaygain.h
new file mode 100644
index 00000000..15be8885
--- /dev/null
+++ b/gst/replaygain/replaygain.h
@@ -0,0 +1,36 @@
+/* GStreamer ReplayGain plugin
+ *
+ * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
+ *
+ * replaygain.h: Plugin providing ReplayGain related elements
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public License
+ * as published by the Free Software Foundation; either version 2.1 of
+ * the License, or (at your option) any later version.
+ * 
+ * This library is distributed in the hope that it will be useful, but
+ * WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
+ * Lesser General Public License for more details.
+ * 
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
+ * 02110-1301 USA
+ */
+
+#ifndef __REPLAYGAIN_H__
+#define __REPLAYGAIN_H__
+
+G_BEGIN_DECLS
+
+/* Reference level (in dBSPL).  The 2001 proposal specifies 83.  This was
+ * changed later in all implementations to 89, which is the new, offical value:
+ * David Robinson acknowledged the change but didn't update the website yet. */
+
+#define RG_REFERENCE_LEVEL 89.
+
+G_END_DECLS
+
+#endif /* __REPLAYGAIN_H__ */
diff --git a/gst/replaygain/rganalysis.h b/gst/replaygain/rganalysis.h
index 39bf9b41..16247361 100644
--- a/gst/replaygain/rganalysis.h
+++ b/gst/replaygain/rganalysis.h
@@ -29,8 +29,6 @@
 
 G_BEGIN_DECLS
 
-#define RG_REFERENCE_LEVEL 89.
-
 typedef struct _RgAnalysisCtx RgAnalysisCtx;
 
 RgAnalysisCtx *rg_analysis_new (void);
diff --git a/tests/check/elements/rganalysis.c b/tests/check/elements/rganalysis.c
index 63e3c720..e8a9db2f 100644
--- a/tests/check/elements/rganalysis.c
+++ b/tests/check/elements/rganalysis.c
@@ -20,77 +20,72 @@
  * 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:
+/* 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.
+ *  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.
+ *  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.
+ *  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.).
+ *  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.
+ *  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.
+ * 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.
+ * 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 */
+/* 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.
+/* 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.
+ * 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. */
+ * 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
 {
@@ -212,11 +207,10 @@ send_eos_event (GstElement * element)
   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. */
+  /* 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);
@@ -251,8 +245,8 @@ poll_eos (GstElement * element)
   gst_object_unref (bus);
 }
 
-/* This also polls for EOS since the TAG message comes right before
- * the end of streams. */
+/* This also polls for EOS since the TAG message comes right before the end of
+ * streams. */
 
 static GstTagList *
 poll_tags (GstElement * element)
@@ -749,14 +743,13 @@ 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
+/* 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)
@@ -1089,11 +1082,10 @@ GST_START_TEST (test_peak_track_album)
 
 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. */
+/* 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)
 {
@@ -1136,8 +1128,8 @@ GST_START_TEST (test_gain_album)
   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. */
+  /* 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--;)
@@ -1268,12 +1260,11 @@ GST_START_TEST (test_forced_separate)
 
 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. */
+/* 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)
 {
@@ -1311,8 +1302,8 @@ GST_START_TEST (test_forced_after_data)
 
 GST_END_TEST;
 
-/* Like test_forced, but *analyze* an album afterwards.  The two tests
- * following this one check the *skipping* of albums. */
+/* 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)
 {
@@ -1441,9 +1432,8 @@ GST_START_TEST (test_forced_album_no_skip)
   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. */
+  /* 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,
@@ -1515,10 +1505,10 @@ GST_START_TEST (test_reference_level)
 {
   GstElement *element = setup_rganalysis ();
   GstTagList *tag_list;
+  gdouble ref_level;
   gint accumulator = 0;
   gint i;
 
-  g_object_set (element, "reference-level", 83., "num-tracks", 2, NULL);
   set_playing_state (element);
 
   for (i = 20; i--;)
@@ -1527,8 +1517,26 @@ GST_START_TEST (test_reference_level)
   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);
+  fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
+          &ref_level) && MATCH_GAIN (ref_level, 89.),
+      "Incorrect reference level tag");
+  gst_tag_list_free (tag_list);
+
+  g_object_set (element, "reference-level", 83., "num-tracks", 2, NULL);
+
+  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);
+  fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
+          &ref_level) && MATCH_GAIN (ref_level, 83.),
+      "Incorrect reference level tag");
   gst_tag_list_free (tag_list);
 
   accumulator = 0;
@@ -1543,6 +1551,9 @@ GST_START_TEST (test_reference_level)
   /* 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.);
+  fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
+          &ref_level) && MATCH_GAIN (ref_level, 83.),
+      "Incorrect reference level tag");
   gst_tag_list_free (tag_list);
 
   cleanup_rganalysis (element);
diff --git a/tests/check/elements/rglimiter.c b/tests/check/elements/rglimiter.c
new file mode 100644
index 00000000..2d4a715b
--- /dev/null
+++ b/tests/check/elements/rglimiter.c
@@ -0,0 +1,238 @@
+/* GStreamer ReplayGain limiter
+ *
+ * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
+ *
+ * rglimiter.c: Unit test for the rglimiter 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
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <math.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;
+
+#define RG_LIMITER_CAPS_TEMPLATE_STRING   \
+  "audio/x-raw-float, "                   \
+  "width = (int) 32, "                    \
+  "endianness = (int) BYTE_ORDER, "       \
+  "channels = (int) [ 1, MAX ], "         \
+  "rate = (int) [ 1, MAX ]"
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (RG_LIMITER_CAPS_TEMPLATE_STRING)
+    );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (RG_LIMITER_CAPS_TEMPLATE_STRING)
+    );
+
+GstElement *
+setup_rglimiter ()
+{
+  GstElement *element;
+  GstBus *bus;
+
+  GST_DEBUG ("setup_rglimiter");
+  element = gst_check_setup_element ("rglimiter");
+  mysrcpad = gst_check_setup_src_pad (element, &srctemplate, NULL);
+  mysinkpad = gst_check_setup_sink_pad (element, &sinktemplate, NULL);
+  gst_pad_set_active (mysrcpad, TRUE);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  return element;
+}
+
+void
+cleanup_rglimiter (GstElement * element)
+{
+  GST_DEBUG ("cleanup_rglimiter");
+
+  g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
+  g_list_free (buffers);
+  buffers = 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 const gfloat test_input[] = {
+  -2.0, -1.0, -0.75, -0.5, -0.25, 0.0, 0.25, 0.5, 0.75, 1.0, 2.0
+};
+static const gfloat test_output[] = {
+  -0.99752737684336523,         /* -2.0  */
+  -0.88079707797788243,         /* -1.0  */
+  -0.7310585786300049,          /* -0.75 */
+  -0.5, -0.25, 0.0, 0.25, 0.5,
+  0.7310585786300049,           /* 0.75 */
+  0.88079707797788243,          /* 1.0  */
+  0.99752737684336523,          /* 2.0  */
+};
+
+static GstBuffer *
+create_test_buffer ()
+{
+  GstBuffer *buf = gst_buffer_new_and_alloc (sizeof (test_input));
+  GstCaps *caps;
+
+  memcpy (GST_BUFFER_DATA (buf), test_input, sizeof (test_input));
+
+  caps = gst_caps_new_simple ("audio/x-raw-float",
+      "rate", G_TYPE_INT, 44100, "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 void
+verify_test_buffer (GstBuffer * buf)
+{
+  gfloat *output = (gfloat *) GST_BUFFER_DATA (buf);
+  gint i;
+
+  fail_unless (GST_BUFFER_SIZE (buf) == sizeof (test_output));
+  for (i = 0; i < G_N_ELEMENTS (test_input); i++)
+    fail_unless (ABS (output[i] - test_output[i]) < 1.e-6,
+        "Incorrect output value %.6f for input %.2f, expected %.6f",
+        output[i], test_input[i], test_output[i]);
+}
+
+/* Start of tests. */
+
+GST_START_TEST (test_no_buffer)
+{
+  GstElement *element = setup_rglimiter ();
+
+  set_playing_state (element);
+
+  cleanup_rglimiter (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_disabled)
+{
+  GstElement *element = setup_rglimiter ();
+  GstBuffer *buf, *out_buf;
+
+  g_object_set (element, "enabled", FALSE, NULL);
+  set_playing_state (element);
+
+  buf = create_test_buffer ();
+  fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
+  fail_unless (g_list_length (buffers) == 1);
+  out_buf = buffers->data;
+  fail_if (out_buf == NULL);
+  buffers = g_list_remove (buffers, out_buf);
+  ASSERT_BUFFER_REFCOUNT (out_buf, "out_buf", 1);
+  fail_unless (buf == out_buf);
+  gst_buffer_unref (out_buf);
+
+  cleanup_rglimiter (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_limiting)
+{
+  GstElement *element = setup_rglimiter ();
+  GstBuffer *buf, *out_buf;
+
+  set_playing_state (element);
+
+  /* Mutable variant. */
+  buf = create_test_buffer ();
+  fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
+  fail_unless (g_list_length (buffers) == 1);
+  out_buf = buffers->data;
+  fail_if (out_buf == NULL);
+  ASSERT_BUFFER_REFCOUNT (out_buf, "out_buf", 1);
+  verify_test_buffer (out_buf);
+
+  /* Immutable variant. */
+  buf = create_test_buffer ();
+  /* Extra ref: */
+  gst_buffer_ref (buf);
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 2);
+  fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
+  fail_unless (g_list_length (buffers) == 2);
+  out_buf = g_list_last (buffers)->data;
+  fail_if (out_buf == NULL);
+  ASSERT_BUFFER_REFCOUNT (out_buf, "out_buf", 1);
+  fail_unless (buf != out_buf);
+  /* Drop our extra ref: */
+  gst_buffer_unref (buf);
+  verify_test_buffer (out_buf);
+
+  cleanup_rglimiter (element);
+}
+
+GST_END_TEST;
+
+Suite *
+rglimiter_suite (void)
+{
+  Suite *s = suite_create ("rglimiter");
+  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_disabled);
+  tcase_add_test (tc_chain, test_limiting);
+
+  return s;
+}
+
+int
+main (int argc, char **argv)
+{
+  gint nf;
+
+  Suite *s = rglimiter_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 --git a/tests/check/elements/rgvolume.c b/tests/check/elements/rgvolume.c
new file mode 100644
index 00000000..658e98ce
--- /dev/null
+++ b/tests/check/elements/rgvolume.c
@@ -0,0 +1,573 @@
+/* GStreamer ReplayGain volume adjustment
+ *
+ * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
+ *
+ * rgvolume.c: Unit test for the rgvolume 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
+ */
+
+#include <gst/check/gstcheck.h>
+
+#include <math.h>
+
+GList *buffers = NULL;
+GList *events = 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;
+
+#define RG_VOLUME_CAPS_TEMPLATE_STRING    \
+  "audio/x-raw-float, "                   \
+  "width = (int) 32, "                    \
+  "endianness = (int) BYTE_ORDER, "       \
+  "channels = (int) [ 1, MAX ], "         \
+  "rate = (int) [ 1, MAX ]"
+
+static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
+    GST_PAD_SINK,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (RG_VOLUME_CAPS_TEMPLATE_STRING)
+    );
+static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
+    GST_PAD_SRC,
+    GST_PAD_ALWAYS,
+    GST_STATIC_CAPS (RG_VOLUME_CAPS_TEMPLATE_STRING)
+    );
+
+/* gstcheck sets up a chain function that appends buffers to a global list.
+ * This is our equivalent of that for event handling. */
+static gboolean
+event_func (GstPad * pad, GstEvent * event)
+{
+  events = g_list_append (events, event);
+
+  return TRUE;
+}
+
+GstElement *
+setup_rgvolume ()
+{
+  GstElement *element;
+
+  GST_DEBUG ("setup_rgvolume");
+  element = gst_check_setup_element ("rgvolume");
+  mysrcpad = gst_check_setup_src_pad (element, &srctemplate, NULL);
+  mysinkpad = gst_check_setup_sink_pad (element, &sinktemplate, NULL);
+
+  /* Capture events, to test tag filtering behavior: */
+  gst_pad_set_event_function (mysinkpad, event_func);
+
+  gst_pad_set_active (mysrcpad, TRUE);
+  gst_pad_set_active (mysinkpad, TRUE);
+
+  return element;
+}
+
+void
+cleanup_rgvolume (GstElement * element)
+{
+  GST_DEBUG ("cleanup_rgvolume");
+
+  g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
+  g_list_free (buffers);
+  buffers = NULL;
+
+  g_list_foreach (events, (GFunc) gst_mini_object_unref, NULL);
+  g_list_free (events);
+  events = NULL;
+
+  gst_pad_set_active (mysrcpad, FALSE);
+  gst_pad_set_active (mysinkpad, FALSE);
+  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
+set_null_state (GstElement * element)
+{
+  fail_unless (gst_element_set_state (element,
+          GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS,
+      "Could not set state to NULL");
+}
+
+static void
+send_eos_event (GstElement * element)
+{
+  GstEvent *event = gst_event_new_eos ();
+
+  fail_unless (g_list_length (events) == 0);
+  fail_unless (gst_pad_push_event (mysrcpad, event),
+      "Pushing EOS event failed");
+  fail_unless (g_list_length (events) == 1);
+  fail_unless (events->data == event);
+  gst_mini_object_unref ((GstMiniObject *) events->data);
+  events = g_list_remove (events, event);
+}
+
+static GstEvent *
+send_tag_event (GstElement * element, GstEvent * event)
+{
+  g_return_val_if_fail (event->type == GST_EVENT_TAG, NULL);
+
+  fail_unless (g_list_length (events) == 0);
+  fail_unless (gst_pad_push_event (mysrcpad, event),
+      "Pushing tag event failed");
+
+  if (g_list_length (events) == 0) {
+    /* Event got filtered out. */
+    event = NULL;
+  } else {
+    GstTagList *tag_list;
+    gdouble dummy;
+
+    event = events->data;
+    events = g_list_remove (events, event);
+
+    fail_unless (event->type == GST_EVENT_TAG);
+    gst_event_parse_tag (event, &tag_list);
+
+    /* The element is supposed to filter out ReplayGain related tags. */
+    fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN, &dummy),
+        "tag event still contains track gain tag");
+    fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK, &dummy),
+        "tag event still contains track peak tag");
+    fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN, &dummy),
+        "tag event still contains album gain tag");
+    fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK, &dummy),
+        "tag event still contains album peak tag");
+  }
+
+  return event;
+}
+
+static GstBuffer *
+test_buffer_new (gfloat value)
+{
+  GstBuffer *buf;
+  GstCaps *caps;
+  gfloat *data;
+  gint i;
+
+  buf = gst_buffer_new_and_alloc (8 * sizeof (gfloat));
+  data = (gfloat *) GST_BUFFER_DATA (buf);
+  for (i = 0; i < 8; i++)
+    data[i] = value;
+
+  caps = gst_caps_from_string ("audio/x-raw-float, "
+      "rate = 8000, channels = 1, endianess = BYTE_ORDER, width = 32");
+  gst_buffer_set_caps (buf, caps);
+  gst_caps_unref (caps);
+
+  ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
+
+  return buf;
+}
+
+#define MATCH_GAIN(g1, g2) ((g1 < g2 + 1e-6) && (g2 < g1 + 1e-6))
+
+static void
+fail_unless_target_gain (GstElement * element, gdouble expected_gain)
+{
+  gdouble prop_gain;
+
+  g_object_get (element, "target-gain", &prop_gain, NULL);
+
+  fail_unless (MATCH_GAIN (prop_gain, expected_gain),
+      "Target gain is %.2f dB, expected %.2f dB", prop_gain, expected_gain);
+}
+
+static void
+fail_unless_result_gain (GstElement * element, gdouble expected_gain)
+{
+  GstBuffer *input_buf, *output_buf;
+  gfloat input_sample, output_sample;
+  gdouble gain, prop_gain;
+  gboolean is_passthrough, expect_passthrough;
+  gint i;
+
+  fail_unless (g_list_length (buffers) == 0);
+
+  input_sample = 1.0;
+  input_buf = test_buffer_new (input_sample);
+
+  /* We keep an extra reference to detect passthrough mode. */
+  gst_buffer_ref (input_buf);
+  /* Pushing steals a reference. */
+  fail_unless (gst_pad_push (mysrcpad, input_buf) == GST_FLOW_OK);
+  gst_buffer_unref (input_buf);
+
+  /* The output buffer ends up on the global buffer list. */
+  fail_unless (g_list_length (buffers) == 1);
+  output_buf = buffers->data;
+  fail_if (output_buf == NULL);
+
+  buffers = g_list_remove (buffers, output_buf);
+  ASSERT_BUFFER_REFCOUNT (output_buf, "output_buf", 1);
+  fail_unless_equals_int (GST_BUFFER_SIZE (output_buf), 8 * sizeof (gfloat));
+
+  output_sample = *((gfloat *) GST_BUFFER_DATA (output_buf));
+
+  fail_if (output_sample == 0.0, "First output sample is zero");
+  for (i = 1; i < 8; i++) {
+    gfloat output = ((gfloat *) GST_BUFFER_DATA (output_buf))[i];
+
+    fail_unless (output_sample == output, "Output samples not uniform");
+  };
+
+  gain = 20. * log10 (output_sample / input_sample);
+  fail_unless (MATCH_GAIN (gain, expected_gain),
+      "Applied gain is %.2f dB, expected %.2f dB", gain, expected_gain);
+  g_object_get (element, "result-gain", &prop_gain, NULL);
+  fail_unless (MATCH_GAIN (prop_gain, expected_gain),
+      "Result gain is %.2f dB, expected %.2f dB", prop_gain, expected_gain);
+
+  is_passthrough = (output_buf == input_buf);
+  expect_passthrough = MATCH_GAIN (expected_gain, +0.00);
+  fail_unless (is_passthrough == expect_passthrough,
+      expect_passthrough
+      ? "Expected operation in passthrough mode"
+      : "Incorrect passthrough behaviour");
+
+  gst_buffer_unref (output_buf);
+}
+
+static void
+fail_unless_gain (GstElement * element, gdouble expected_gain)
+{
+  fail_unless_target_gain (element, expected_gain);
+  fail_unless_result_gain (element, expected_gain);
+}
+
+/* Start of tests. */
+
+GST_START_TEST (test_no_buffer)
+{
+  GstElement *element = setup_rgvolume ();
+
+  set_playing_state (element);
+  set_null_state (element);
+  set_playing_state (element);
+  send_eos_event (element);
+
+  cleanup_rgvolume (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_events)
+{
+  GstElement *element = setup_rgvolume ();
+  GstEvent *event;
+  GstEvent *new_event;
+  GstTagList *tag_list;
+  gchar *artist;
+
+  set_playing_state (element);
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_TRACK_GAIN, +4.95, GST_TAG_TRACK_PEAK, 0.59463,
+      GST_TAG_ALBUM_GAIN, -1.54, GST_TAG_ALBUM_PEAK, 0.693415,
+      GST_TAG_ARTIST, "Foobar", NULL);
+  event = gst_event_new_tag (tag_list);
+  new_event = send_tag_event (element, event);
+  /* Expect the element to modify the writable event. */
+  fail_unless (event == new_event, "Writable tag event not reused");
+  gst_event_parse_tag (new_event, &tag_list);
+  fail_unless (gst_tag_list_get_string (tag_list, GST_TAG_ARTIST, &artist));
+  fail_unless (g_str_equal (artist, "Foobar"));
+  g_free (artist);
+  gst_event_unref (new_event);
+
+  /* Same as above, but with a non-writable event. */
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_TRACK_GAIN, +4.95, GST_TAG_TRACK_PEAK, 0.59463,
+      GST_TAG_ALBUM_GAIN, -1.54, GST_TAG_ALBUM_PEAK, 0.693415,
+      GST_TAG_ARTIST, "Foobar", NULL);
+  event = gst_event_new_tag (tag_list);
+  /* Holding an extra ref makes the event unwritable: */
+  gst_event_ref (event);
+  new_event = send_tag_event (element, event);
+  fail_unless (event != new_event, "Unwritable tag event reused");
+  gst_event_parse_tag (new_event, &tag_list);
+  fail_unless (gst_tag_list_get_string (tag_list, GST_TAG_ARTIST, &artist));
+  fail_unless (g_str_equal (artist, "Foobar"));
+  g_free (artist);
+  gst_event_unref (event);
+  gst_event_unref (new_event);
+
+  cleanup_rgvolume (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_simple)
+{
+  GstElement *element = setup_rgvolume ();
+  GstTagList *tag_list;
+
+  g_object_set (element, "album-mode", FALSE, "headroom", +0.00,
+      "pre-amp", -6.00, "fallback-gain", +1.23, NULL);
+  set_playing_state (element);
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_TRACK_GAIN, -3.45, GST_TAG_TRACK_PEAK, 1.0,
+      GST_TAG_ALBUM_GAIN, +2.09, GST_TAG_ALBUM_PEAK, 1.0, NULL);
+  fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
+  fail_unless_gain (element, -9.45);    /* pre-amp + track gain */
+  send_eos_event (element);
+
+  g_object_set (element, "album-mode", TRUE, NULL);
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_TRACK_GAIN, -3.45, GST_TAG_TRACK_PEAK, 1.0,
+      GST_TAG_ALBUM_GAIN, +2.09, GST_TAG_ALBUM_PEAK, 1.0, NULL);
+  fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
+  fail_unless_gain (element, -3.91);    /* pre-amp + album gain */
+
+  /* Switching back to track mode in the middle of a stream: */
+  g_object_set (element, "album-mode", FALSE, NULL);
+  fail_unless_gain (element, -9.45);    /* pre-amp + track gain */
+  send_eos_event (element);
+
+  cleanup_rgvolume (element);
+}
+
+GST_END_TEST;
+
+/* If there are no gain tags at all, the fallback gain is used. */
+
+GST_START_TEST (test_fallback_gain)
+{
+  GstElement *element = setup_rgvolume ();
+  GstTagList *tag_list;
+
+  /* First some track where fallback does _not_ apply. */
+
+  g_object_set (element, "album-mode", FALSE, "headroom", 10.00,
+      "pre-amp", -6.00, "fallback-gain", -3.00, NULL);
+  set_playing_state (element);
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_TRACK_GAIN, +3.5, GST_TAG_TRACK_PEAK, 1.0,
+      GST_TAG_ALBUM_GAIN, -0.5, GST_TAG_ALBUM_PEAK, 1.0, NULL);
+  fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
+  fail_unless_gain (element, -2.50);    /* pre-amp + track gain */
+  send_eos_event (element);
+
+  /* Now a track completely missing tags. */
+
+  fail_unless_gain (element, -9.00);    /* pre-amp + fallback-gain */
+
+  /* Changing the fallback gain in the middle of a stream, going to pass-through
+   * mode: */
+  g_object_set (element, "fallback-gain", +6.00, NULL);
+  fail_unless_gain (element, +0.00);    /* pre-amp + fallback-gain */
+  send_eos_event (element);
+
+  /* Verify that result gain is set to +0.00 with pre-amp + fallback-gain >
+   * +0.00 and no headroom. */
+
+  g_object_set (element, "fallback-gain", +12.00, "headroom", +0.00, NULL);
+  fail_unless_target_gain (element, +6.00);     /* pre-amp + fallback-gain */
+  fail_unless_result_gain (element, +0.00);
+  send_eos_event (element);
+
+  cleanup_rgvolume (element);
+}
+
+GST_END_TEST;
+
+/* If album gain is to be preferred but not available, the track gain is to be
+ * taken instead. */
+
+GST_START_TEST (test_fallback_track)
+{
+  GstElement *element = setup_rgvolume ();
+  GstTagList *tag_list;
+
+  g_object_set (element, "album-mode", TRUE, "headroom", +0.00,
+      "pre-amp", -6.00, "fallback-gain", +1.23, NULL);
+  set_playing_state (element);
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_TRACK_GAIN, +2.11, GST_TAG_TRACK_PEAK, 1.0, NULL);
+  fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
+  fail_unless_gain (element, -3.89);    /* pre-amp + track gain */
+
+  send_eos_event (element);
+
+  cleanup_rgvolume (element);
+}
+
+GST_END_TEST;
+
+/* If track gain is to be preferred but not available, the album gain is to be
+ * taken instead. */
+
+GST_START_TEST (test_fallback_album)
+{
+  GstElement *element = setup_rgvolume ();
+  GstTagList *tag_list;
+
+  g_object_set (element, "album-mode", FALSE, "headroom", +0.00,
+      "pre-amp", -6.00, "fallback-gain", +1.23, NULL);
+  set_playing_state (element);
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_ALBUM_GAIN, +3.73, GST_TAG_ALBUM_PEAK, 1.0, NULL);
+  fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
+  fail_unless_gain (element, -2.27);    /* pre-amp + album gain */
+
+  send_eos_event (element);
+
+  cleanup_rgvolume (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_headroom)
+{
+  GstElement *element = setup_rgvolume ();
+  GstTagList *tag_list;
+
+  g_object_set (element, "album-mode", FALSE, "headroom", +0.00,
+      "pre-amp", +0.00, "fallback-gain", +1.23, NULL);
+  set_playing_state (element);
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_TRACK_GAIN, +3.50, GST_TAG_TRACK_PEAK, 1.0, NULL);
+  fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
+  fail_unless_target_gain (element, +3.50);     /* pre-amp + track gain */
+  fail_unless_result_gain (element, +0.00);
+  send_eos_event (element);
+
+  g_object_set (element, "headroom", +2.00, NULL);
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_TRACK_GAIN, +9.18, GST_TAG_TRACK_PEAK, 0.687149, NULL);
+  fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
+  fail_unless_target_gain (element, +9.18);     /* pre-amp + track gain */
+  /* Result is 20. * log10 (1. / peak) + headroom. */
+  fail_unless_result_gain (element, 5.2589816238303335);
+  send_eos_event (element);
+
+  g_object_set (element, "album-mode", TRUE, NULL);
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_ALBUM_GAIN, +5.50, GST_TAG_ALBUM_PEAK, 1.0, NULL);
+  fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
+  fail_unless_target_gain (element, +5.50);     /* pre-amp + album gain */
+  fail_unless_result_gain (element, +2.00);     /* headroom */
+  send_eos_event (element);
+
+  cleanup_rgvolume (element);
+}
+
+GST_END_TEST;
+
+GST_START_TEST (test_reference_level)
+{
+  GstElement *element = setup_rgvolume ();
+  GstTagList *tag_list;
+
+  g_object_set (element,
+      "album-mode", FALSE,
+      "headroom", +0.00, "pre-amp", +0.00, "fallback-gain", +1.23, NULL);
+  set_playing_state (element);
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_TRACK_GAIN, 0.00, GST_TAG_TRACK_PEAK, 0.2,
+      GST_TAG_REFERENCE_LEVEL, 83., NULL);
+  fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
+  /* Because our authorative reference is 89 dB, we bump it up by +6 dB. */
+  fail_unless_gain (element, +6.00);    /* pre-amp + track gain */
+  send_eos_event (element);
+
+  g_object_set (element, "album-mode", TRUE, NULL);
+
+  /* Same as above, but with album gain. */
+
+  tag_list = gst_tag_list_new ();
+  gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
+      GST_TAG_TRACK_GAIN, 1.23, GST_TAG_TRACK_PEAK, 0.1,
+      GST_TAG_ALBUM_GAIN, 0.00, GST_TAG_ALBUM_PEAK, 0.2,
+      GST_TAG_REFERENCE_LEVEL, 83., NULL);
+  fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
+  fail_unless_gain (element, +6.00);    /* pre-amp + album gain */
+
+  cleanup_rgvolume (element);
+}
+
+GST_END_TEST;
+
+Suite *
+rgvolume_suite (void)
+{
+  Suite *s = suite_create ("rgvolume");
+  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_events);
+  tcase_add_test (tc_chain, test_simple);
+  tcase_add_test (tc_chain, test_fallback_gain);
+  tcase_add_test (tc_chain, test_fallback_track);
+  tcase_add_test (tc_chain, test_fallback_album);
+  tcase_add_test (tc_chain, test_headroom);
+  tcase_add_test (tc_chain, test_reference_level);
+
+  return s;
+}
+
+int
+main (int argc, char **argv)
+{
+  gint nf;
+
+  Suite *s = rgvolume_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;
+}
author	René Stadler <mail@renestadler.de>	2007-05-19 10:01:45 +0000
committer	Tim-Philipp Müller <tim@centricular.net>	2007-05-19 10:01:45 +0000
commit	4e45e0a2690605843115abbbde06ca00a8da8584 (patch)
tree	addb448253cd26656ecf7bba66e87f73f6ff0f61
parent	fc99abef7f1ca971b919b433e87ed46cbd644aae (diff)