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
/**
* 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>
* </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>
*/
#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
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);
filter->ctx = rg_analysis_new ();
filter->analyze = NULL;
- GST_DEBUG_OBJECT (filter, "Started");
+ GST_LOG_OBJECT (filter, "started");
return TRUE;
}
{
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);
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)
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;
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;
{
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),
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;
}
rg_analysis_destroy (filter->ctx);
filter->ctx = NULL;
- GST_DEBUG_OBJECT (filter, "Stopped");
+ GST_LOG_OBJECT (filter, "stopped");
return TRUE;
}
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;
}
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
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);
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)
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) {
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) {
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);
GstBaseTransformClass parent_class;
};
+GType gst_rg_analysis_get_type (void);
+
G_END_DECLS
#endif /* __GST_RG_ANALYSIS_H__ */
--- /dev/null
+/* 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;
+}
--- /dev/null
+/* 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__ */
--- /dev/null
+/* 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;
+ }
+}
--- /dev/null
+/* 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__ */
--- /dev/null
+/* 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);
--- /dev/null
+/* 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__ */
G_BEGIN_DECLS
-#define RG_REFERENCE_LEVEL 89.
-
typedef struct _RgAnalysisCtx RgAnalysisCtx;
RgAnalysisCtx *rg_analysis_new (void);
* 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
{
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);
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)
/* 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)
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)
{
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--;)
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)
{
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)
{
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,
{
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--;)
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;
/* 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);
--- /dev/null
+/* 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;
+}
--- /dev/null
+/* 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;
+}
projects / platform / upstream / gstreamer.git / commitdiff