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Add replaygain playback elements (#412710).

Browse source 
Original commit message from CVS:
Patch by: René Stadler <mail at renestadler de>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-replaygain.xml:
* gst/replaygain/Makefile.am:
* gst/replaygain/gstrganalysis.c: (gst_rg_analysis_class_init),
(gst_rg_analysis_start), (gst_rg_analysis_set_caps),
(gst_rg_analysis_transform_ip), (gst_rg_analysis_event),
(gst_rg_analysis_stop), (gst_rg_analysis_handle_tags),
(gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result),
(gst_rg_analysis_album_result):
* gst/replaygain/gstrganalysis.h:
* gst/replaygain/gstrglimiter.c: (gst_rg_limiter_base_init),
(gst_rg_limiter_class_init), (gst_rg_limiter_init),
(gst_rg_limiter_set_property), (gst_rg_limiter_get_property),
(gst_rg_limiter_transform_ip):
* gst/replaygain/gstrglimiter.h:
* gst/replaygain/gstrgvolume.c: (gst_rg_volume_base_init),
(gst_rg_volume_class_init), (gst_rg_volume_init),
(gst_rg_volume_set_property), (gst_rg_volume_get_property),
(gst_rg_volume_dispose), (gst_rg_volume_change_state),
(gst_rg_volume_sink_event), (gst_rg_volume_tag_event),
(gst_rg_volume_reset), (gst_rg_volume_update_gain),
(gst_rg_volume_determine_gain):
* gst/replaygain/gstrgvolume.h:
* gst/replaygain/replaygain.c: (plugin_init):
* gst/replaygain/replaygain.h:
* gst/replaygain/rganalysis.h:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/rganalysis.c: (send_eos_event),
(GST_START_TEST):
* tests/check/elements/rglimiter.c: (setup_rglimiter),
(cleanup_rglimiter), (set_playing_state), (create_test_buffer),
(verify_test_buffer), (GST_START_TEST), (rglimiter_suite), (main):
* tests/check/elements/rgvolume.c: (event_func), (setup_rgvolume),
(cleanup_rgvolume), (set_playing_state), (set_null_state),
(send_eos_event), (send_tag_event), (test_buffer_new),
(fail_unless_target_gain), (fail_unless_result_gain),
(fail_unless_gain), (GST_START_TEST), (rgvolume_suite), (main):
Add replaygain playback elements (#412710).
This commit is contained in:
René Stadler 2007-05-19 10:01:45 +00:00 committed by Tim-Philipp Müller
parent 8d971fb7fa
commit 1645110f02
20 changed files with 2302 additions and 242 deletions
Show stats Download patch file Download diff file Expand all files Collapse all files

46
ChangeLog
View file

@ -1,3 +1,49 @@
2007-05-19 Tim-Philipp Müller <tim at centricular dot net>
Patch by: René Stadler <mail at renestadler de>
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/inspect/plugin-replaygain.xml:
* gst/replaygain/Makefile.am:
* gst/replaygain/gstrganalysis.c: (gst_rg_analysis_class_init),
(gst_rg_analysis_start), (gst_rg_analysis_set_caps),
(gst_rg_analysis_transform_ip), (gst_rg_analysis_event),
(gst_rg_analysis_stop), (gst_rg_analysis_handle_tags),
(gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result),
(gst_rg_analysis_album_result):
* gst/replaygain/gstrganalysis.h:
* gst/replaygain/gstrglimiter.c: (gst_rg_limiter_base_init),
(gst_rg_limiter_class_init), (gst_rg_limiter_init),
(gst_rg_limiter_set_property), (gst_rg_limiter_get_property),
(gst_rg_limiter_transform_ip):
* gst/replaygain/gstrglimiter.h:
* gst/replaygain/gstrgvolume.c: (gst_rg_volume_base_init),
(gst_rg_volume_class_init), (gst_rg_volume_init),
(gst_rg_volume_set_property), (gst_rg_volume_get_property),
(gst_rg_volume_dispose), (gst_rg_volume_change_state),
(gst_rg_volume_sink_event), (gst_rg_volume_tag_event),
(gst_rg_volume_reset), (gst_rg_volume_update_gain),
(gst_rg_volume_determine_gain):
* gst/replaygain/gstrgvolume.h:
* gst/replaygain/replaygain.c: (plugin_init):
* gst/replaygain/replaygain.h:
* gst/replaygain/rganalysis.h:
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/rganalysis.c: (send_eos_event),
(GST_START_TEST):
* tests/check/elements/rglimiter.c: (setup_rglimiter),
(cleanup_rglimiter), (set_playing_state), (create_test_buffer),
(verify_test_buffer), (GST_START_TEST), (rglimiter_suite), (main):
* tests/check/elements/rgvolume.c: (event_func), (setup_rgvolume),
(cleanup_rgvolume), (set_playing_state), (set_null_state),
(send_eos_event), (send_tag_event), (test_buffer_new),
(fail_unless_target_gain), (fail_unless_result_gain),
(fail_unless_gain), (GST_START_TEST), (rgvolume_suite), (main):
Add replaygain playback elements (#412710).
2007-05-18 Jan Schmidt <thaytan@mad.scientist.com>
* sys/glsink/glimagesink.c: (gst_glimage_sink_init_display):

2
docs/plugins/Makefile.am
View file

@ -99,6 +99,8 @@ EXTRA_HFILES = \
$(top_srcdir)/ext/wavpack/gstwavpackparse.h \
$(top_srcdir)/gst/qtdemux/qtdemux.h \
$(top_srcdir)/gst/replaygain/gstrganalysis.h \
$(top_srcdir)/gst/replaygain/gstrglimiter.h \
$(top_srcdir)/gst/replaygain/gstrgvolume.h \
$(top_srcdir)/gst/videocrop/gstvideocrop.h
# Images to copy into HTML directory.

2
docs/plugins/gst-plugins-bad-plugins-docs.sgml
View file

@ -20,6 +20,8 @@
<xi:include href="xml/element-qtdemux.xml" />
<xi:include href="xml/element-osxvideosink.xml" />
<xi:include href="xml/element-rganalysis.xml" />
<xi:include href="xml/element-rglimiter.xml" />
<xi:include href="xml/element-rgvolume.xml" />
<xi:include href="xml/element-sdlaudiosink.xml" />
<xi:include href="xml/element-sdlvideosink.xml" />
<xi:include href="xml/element-trm.xml" />

16
docs/plugins/gst-plugins-bad-plugins-sections.txt
View file

@ -55,6 +55,22 @@ GstRgAnalysis
GstRgAnalysisClass
</SECTION>
<SECTION>
<FILE>element-rglimiter</FILE>
GstRgLimiter
<TITLE>rglimiter</TITLE>
<SUBSECTION Standard>
GstRgLimiterClass
</SECTION>
<SECTION>
<FILE>element-rgvolume</FILE>
GstRgVolume
<TITLE>rgvolume</TITLE>
<SUBSECTION Standard>
GstRgVolumeClass
</SECTION>
<SECTION>
<FILE>element-sdlaudiosink</FILE>
GstSDLAudioSink

16
docs/plugins/inspect/plugin-replaygain.xml
View file

@ -1,6 +1,6 @@
<plugin>
<name>replaygain</name>
<description>ReplayGain analysis</description>
<description>ReplayGain volume normalization</description>
<filename>../../gst/replaygain/.libs/libgstreplaygain.so</filename>
<basename>libgstreplaygain.so</basename>
<version>0.10.4.1</version>
@ -16,5 +16,19 @@
<description>Perform the ReplayGain analysis</description>
<author>René Stadler &lt;mail@renestadler.de&gt;</author>
</element>
<element>
<name>rglimiter</name>
<longname>ReplayGain limiter</longname>
<class>Filter/Effect/Audio</class>
<description>Apply signal compression to raw audio data</description>
<author>René Stadler &lt;mail@renestadler.de&gt;</author>
</element>
<element>
<name>rgvolume</name>
<longname>ReplayGain volume</longname>
<class>Filter/Effect/Audio</class>
<description>Apply ReplayGain volume adjustment</description>
<author>René Stadler &lt;mail@renestadler.de&gt;</author>
</element>
</elements>
</plugin>

12
gst/replaygain/Makefile.am
View file

@ -2,12 +2,20 @@ plugin_LTLIBRARIES = libgstreplaygain.la
libgstreplaygain_la_SOURCES = \
gstrganalysis.c \
gstrglimiter.c \
gstrgvolume.c \
replaygain.c \
rganalysis.c
libgstreplaygain_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS)
libgstreplaygain_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(LIBM)
libgstreplaygain_la_CFLAGS = \
$(GST_CFLAGS) $(GST_BASE_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
libgstreplaygain_la_LIBADD = \
$(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstpbutils-0.10 $(LIBM)
libgstreplaygain_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
# headers we need but don't want installed
noinst_HEADERS = \
gstrganalysis.h \
gstrglimiter.h \
gstrgvolume.h \
replaygain.h \
rganalysis.h

298
gst/replaygain/gstrganalysis.c
View file

@ -22,103 +22,29 @@
/**
* SECTION:element-rganalysis
* @see_also: <link linkend="GstRgVolume">rgvolume</link>
*
* <refsect2>
* <para>
* GstRgAnalysis analyzes raw audio sample data in accordance with the
* proposed <ulink url="http://replaygain.org">ReplayGain
* standard</ulink> for calculating the ideal replay gain for music
* tracks and albums. The element is designed as a pass-through
* filter that never modifies any data. As it receives an EOS event,
* it finalizes the ongoing analysis and generates a tag list
* containing the results. It is sent downstream with a TAG event and
* posted on the message bus with a TAG message. The EOS event is
* forwarded as normal afterwards. Result tag lists at least contain
* the tags #GST_TAG_TRACK_GAIN and #GST_TAG_TRACK_PEAK.
* This element analyzes raw audio sample data in accordance with the proposed
* <ulink url="http://replaygain.org">ReplayGain standard</ulink> for
* calculating the ideal replay gain for music tracks and albums. The element
* is designed as a pass-through filter that never modifies any data. As it
* receives an EOS event, it finalizes the ongoing analysis and generates a tag
* list containing the results. It is sent downstream with a tag event and
* posted on the message bus with a tag message. The EOS event is forwarded as
* normal afterwards. Result tag lists at least contain the tags
* #GST_TAG_TRACK_GAIN, #GST_TAG_TRACK_PEAK and #GST_TAG_REFERENCE_LEVEL.
* </para>
* <title>Album processing</title>
* <para>
* Analyzing several streams sequentially and assigning them a common
* result gain is known as "album processing". If this gain is used
* during playback (by switching to "album mode"), all tracks receive
* the same amplification. This keeps the relative volume levels
* between the tracks intact. To enable this, set the <link
* linkend="GstRgAnalysis--num-tracks">num-tracks</link> property to
* the number of streams that will be processed as album tracks.
* Every time an EOS event is received, the value of this property
* will be decremented by one. As it reaches zero, it is assumed that
* the last track of the album finished. The tag list for the final
* stream will contain the additional tags #GST_TAG_ALBUM_GAIN and
* #GST_TAG_ALBUM_PEAK. All other streams just get the two track tags
* posted because the values for the album tags are not known before
* all tracks are analyzed. Applications need to make sure that the
* album gain and peak values are also associated with the other
* tracks when storing the results. It is thus a bit more complex to
* implement, but should not be avoided since the album gain is
* generally more valuable for use during playback than the track
* gain.
* </para>
* <title>Skipping processing</title>
* <para>
* For assisting transcoder/converter applications, the element can
* silently skip the processing of streams that already contain the
* necessary meta data tags. Data will flow as usual but the element
* will not consume CPU time and will not generate result tags. To
* enable possible skipping, set the <link
* linkend="GstRgAnalysis--forced">forced</link> property to #FALSE.
* If used in conjunction with album processing, the element will skip
* the number of remaining album tracks if a full set of tags is found
* for the first track. If a subsequent track of the album is missing
* tags, processing cannot start again. If this is undesired, your
* application has to scan all files beforehand and enable forcing of
* processing if needed.
* </para>
* <title>Tips</title>
* <itemizedlist>
* <listitem><para>
* Because the generated metadata tags become available at the end of
* streams, downstream muxer and encoder elements are normally unable
* to save them in their output since they generally save metadata in
* the file header. Therefore, it is often necessary that
* applications read the results in a bus event handler for the tag
* message. Obtaining the values this way is always needed for album
* processing since the album gain and peak values need to be
* associated with all tracks of an album, not just the last one.
* </para></listitem>
* <listitem><para>
* To perform album processing, the element has to preserve data
* between streams. This cannot survive a state change to the NULL or
* READY state. If you change your pipeline's state to NULL or READY
* between tracks, lock the rganalysis element's state using
* gst_element_set_locked_state() when it is in PAUSED or PLAYING. As
* with any other element, don't forget to unlock it again and set it
* to the NULL state before dropping the last reference.
* </para></listitem>
* <listitem><para>
* If the total number of album tracks is unknown beforehand, set the
* num-tracks property to some large value like #G_MAXINT (or set it
* to >= 2 before each track starts). Before the last track ends, set
* the property value to 1.
* </para></listitem>
* </itemizedlist>
* <title>Compliance</title>
* <para>
* Analyzing the ReplayGain pink noise reference waveform will compute
* a result of +6.00dB instead of the expected 0.00dB because the
* default reference level is 89dB. To obtain values as lined out in
* the original proposal of ReplayGain, set the <link
* linkend="GstRgAnalysis--reference-level">reference-level</link>
* property to 83. Almost all software uses 89dB as a reference
* however, which works against the tendency of the algorithm to
* advise to drastically lower the volume of music with a highly
* compressed dynamic range and high average output levels. This
* tendency is normally to be fought during playback (if wanted), by
* using a default pre-amp value of at least +6.00dB. At one point,
* the majority of analyzer implementations switched to 89dB which
* moved this adjustment to the analyzing/metadata writing process.
* This change has been acknowledged by the author of the ReplayGain
* proposal, however at the time of this writing, the webpage is still
* not updated.
* Because the generated metadata tags become available at the end of streams,
* downstream muxer and encoder elements are normally unable to save them in
* their output since they generally save metadata in the file header.
* Therefore, it is often necessary that applications read the results in a bus
* event handler for the tag message. Obtaining the values this way is always
* needed for <link linkend="GstRgAnalysis--num-tracks">album processing</link>
* since the album gain and peak values need to be associated with all tracks of
* an album, not just the last one.
* </para>
* <title>Example launch lines</title>
* <para>Analyze a simple test waveform:</para>
@ -127,18 +53,26 @@
* </programlisting>
* <para>Analyze a given file:</para>
* <programlisting>
* gst-launch -t filesrc location="Some file.ogg" ! decodebin ! audioconvert ! audioresample ! rganalysis ! fakesink
* gst-launch -t filesrc location="Some file.ogg" ! decodebin \
* ! audioconvert ! audioresample ! rganalysis ! fakesink
* </programlisting>
* <para>Analyze the pink noise reference file:</para>
* <programlisting>
* gst-launch -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav ! wavparse ! rganalysis ! fakesink
* gst-launch -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav \
* ! wavparse ! rganalysis ! fakesink
* </programlisting>
* <para>
* The above launch line yields a result gain of +6 dB (instead of the expected
* +0 dB). This is not in error, refer to the <link
* linkend="GstRgAnalysis--reference-level">reference-level</link> property
* documentation for more information.
* </para>
* <title>Acknowledgements</title>
* <para>
* This element is based on code used in the <ulink
* url="http://sjeng.org/vorbisgain.html">vorbisgain</ulink> program
* and many others. The relevant parts are copyrighted by David
* Robinson, Glen Sawyer and Frank Klemm.
* url="http://sjeng.org/vorbisgain.html">vorbisgain</ulink> program and many
* others. The relevant parts are copyrighted by David Robinson, Glen Sawyer
* and Frank Klemm.
* </para>
* </refsect2>
*/
@ -147,11 +81,11 @@
#include <config.h>
#endif
#include <string.h>
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include "gstrganalysis.h"
#include "replaygain.h"
GST_DEBUG_CATEGORY_STATIC (gst_rg_analysis_debug);
#define GST_CAT_DEFAULT gst_rg_analysis_debug
@ -254,18 +188,93 @@ gst_rg_analysis_class_init (GstRgAnalysisClass * klass)
gobject_class->set_property = gst_rg_analysis_set_property;
gobject_class->get_property = gst_rg_analysis_get_property;
/**
* GstRgAnalysis:num-tracks:
*
* Number of remaining album tracks.
*
* Analyzing several streams sequentially and assigning them a common result
* gain is known as "album processing". If this gain is used during playback
* (by switching to "album mode"), all tracks of an album receive the same
* amplification. This keeps the relative volume levels between the tracks
* intact. To enable this, set this property to the number of streams that
* will be processed as album tracks.
*
* Every time an EOS event is received, the value of this property is
* decremented by one. As it reaches zero, it is assumed that the last track
* of the album finished. The tag list for the final stream will contain the
* additional tags #GST_TAG_ALBUM_GAIN and #GST_TAG_ALBUM_PEAK. All other
* streams just get the two track tags posted because the values for the album
* tags are not known before all tracks are analyzed. Applications need to
* ensure that the album gain and peak values are also associated with the
* other tracks when storing the results.
*
* If the total number of album tracks is unknown beforehand, just ensure that
* the value is greater than 1 before each track starts. Then before the end
* of the last track, set it to the value 1.
*
* To perform album processing, the element has to preserve data between
* streams. This cannot survive a state change to the NULL or READY state.
* If you change your pipeline's state to NULL or READY between tracks, lock
* the element's state using gst_element_set_locked_state() when it is in
* PAUSED or PLAYING.
*/
g_object_class_install_property (gobject_class, PROP_NUM_TRACKS,
g_param_spec_int ("num-tracks", "Number of album tracks",
"Number of remaining tracks in the album",
0, G_MAXINT, 0, G_PARAM_READWRITE));
"Number of remaining album tracks", 0, G_MAXINT, 0,
G_PARAM_READWRITE));
/**
* GstRgAnalysis:forced:
*
* Whether to analyze streams even when ReplayGain tags exist.
*
* For assisting transcoder/converter applications, the element can silently
* skip the processing of streams that already contain the necessary tags.
* Data will flow as usual but the element will not consume CPU time and will
* not generate result tags. To enable possible skipping, set this property
* to #FALSE.
*
* If used in conjunction with <link linkend="GstRgAnalysis--num-tracks">album
* processing</link>, the element will skip the number of remaining album
* tracks if a full set of tags is found for the first track. If a subsequent
* track of the album is missing tags, processing cannot start again. If this
* is undesired, the application has to scan all files beforehand and enable
* forcing of processing if needed.
*/
g_object_class_install_property (gobject_class, PROP_FORCED,
g_param_spec_boolean ("forced", "Force processing",
"Analyze streams even when ReplayGain tags exist",
g_param_spec_boolean ("forced", "Forced",
"Analyze even if ReplayGain tags exist",
FORCED_DEFAULT, G_PARAM_READWRITE));
/**
* GstRgAnalysis:reference-level:
*
* Reference level [dB].
*
* Analyzing the ReplayGain pink noise reference waveform computes a result of
* +6 dB instead of the expected 0 dB. This is because the default reference
* level is 89 dB. To obtain values as lined out in the original proposal of
* ReplayGain, set this property to 83.
*
* Almost all software uses 89 dB as a reference however, and this value has
* become the new official value. That is to say, while the change has been
* acclaimed by the author of the ReplayGain proposal, the <ulink
* url="http://replaygain.org">webpage</ulink> is still outdated at the time
* of this writing.
*
* The value was changed because the original proposal recommends a default
* pre-amp value of +6 dB for playback. This seemed a bit odd, as it means
* that the algorithm has the general tendency to produce adjustment values
* that are 6 dB too low. Bumping the reference level by 6 dB compensated for
* this.
*
* The problem of the reference level being ambiguous for lack of concise
* standardization is to be solved by adopting the #GST_TAG_REFERENCE_LEVEL
* tag, which allows to store the used value alongside the gain values.
*/
g_object_class_install_property (gobject_class, PROP_REFERENCE_LEVEL,
g_param_spec_double ("reference-level", "Reference level",
"Reference level in dB (83.0 for original proposal)",
0.0, G_MAXDOUBLE, RG_REFERENCE_LEVEL, G_PARAM_READWRITE));
"Reference level [dB]", 0.0, 150., RG_REFERENCE_LEVEL,
G_PARAM_READWRITE));
trans_class = (GstBaseTransformClass *) klass;
trans_class->start = GST_DEBUG_FUNCPTR (gst_rg_analysis_start);
@ -346,7 +355,7 @@ gst_rg_analysis_start (GstBaseTransform * base)
filter->ctx = rg_analysis_new ();
filter->analyze = NULL;
GST_DEBUG_OBJECT (filter, "Started");
GST_LOG_OBJECT (filter, "started");
return TRUE;
}
@ -357,7 +366,7 @@ gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
{
GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
GstStructure *structure;
const gchar *mime_type;
const gchar *name;
gint n_channels, sample_rate, sample_bit_size, sample_size;
g_return_val_if_fail (filter->ctx != NULL, FALSE);
@ -367,7 +376,7 @@ gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
in_caps, out_caps);
structure = gst_caps_get_structure (in_caps, 0);
mime_type = gst_structure_get_name (structure);
name = gst_structure_get_name (structure);
if (!gst_structure_get_int (structure, "width", &sample_bit_size)
|| !gst_structure_get_int (structure, "channels", &n_channels)
@ -381,7 +390,7 @@ gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
goto invalid_format;
sample_size = sample_bit_size / 8;
if (strcmp (mime_type, "audio/x-raw-float") == 0) {
if (g_str_equal (name, "audio/x-raw-float")) {
if (sample_size != sizeof (gfloat))
goto invalid_format;
@ -398,7 +407,7 @@ gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
else
goto invalid_format;
} else if (strcmp (mime_type, "audio/x-raw-int") == 0) {
} else if (g_str_equal (name, "audio/x-raw-int")) {
if (sample_size != sizeof (gint16))
goto invalid_format;
@ -437,13 +446,13 @@ gst_rg_analysis_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
g_return_val_if_fail (filter->ctx != NULL, GST_FLOW_ERROR);
g_return_val_if_fail (filter->analyze != NULL, GST_FLOW_ERROR);
g_return_val_if_fail (filter->ctx != NULL, GST_FLOW_WRONG_STATE);
g_return_val_if_fail (filter->analyze != NULL, GST_FLOW_NOT_NEGOTIATED);
if (filter->skip)
return GST_FLOW_OK;
GST_DEBUG_OBJECT (filter, "Processing buffer of size %u",
GST_LOG_OBJECT (filter, "processing buffer of size %u",
GST_BUFFER_SIZE (buf));
filter->analyze (filter->ctx, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
@ -463,11 +472,11 @@ gst_rg_analysis_event (GstBaseTransform * base, GstEvent * event)
case GST_EVENT_EOS:
{
GST_DEBUG_OBJECT (filter, "Received EOS event");
GST_LOG_OBJECT (filter, "received EOS event");
gst_rg_analysis_handle_eos (filter);
GST_DEBUG_OBJECT (filter, "Passing on EOS event");
GST_LOG_OBJECT (filter, "passing on EOS event");
break;
}
@ -498,7 +507,7 @@ gst_rg_analysis_stop (GstBaseTransform * base)
rg_analysis_destroy (filter->ctx);
filter->ctx = NULL;
GST_DEBUG_OBJECT (filter, "Stopped");
GST_LOG_OBJECT (filter, "stopped");
return TRUE;
}
@ -514,13 +523,13 @@ gst_rg_analysis_handle_tags (GstRgAnalysis * filter,
filter->ignore_tags = FALSE;
if (filter->skip && album_processing) {
GST_INFO_OBJECT (filter, "Ignoring TAG event: Skipping album");
GST_DEBUG_OBJECT (filter, "ignoring tag event: skipping album");
return;
} else if (filter->skip) {
GST_INFO_OBJECT (filter, "Ignoring TAG event: Skipping track");
GST_DEBUG_OBJECT (filter, "ignoring tag event: skipping track");
return;
} else if (filter->ignore_tags) {
GST_INFO_OBJECT (filter, "Ignoring TAG event: Cannot skip anyways");
GST_DEBUG_OBJECT (filter, "ignoring tag event: cannot skip anyways");
return;
}
@ -534,30 +543,31 @@ gst_rg_analysis_handle_tags (GstRgAnalysis * filter,
GST_TAG_ALBUM_PEAK, &dummy);
if (!(filter->has_track_gain && filter->has_track_peak)) {
GST_INFO_OBJECT (filter, "Track tags not complete yet");
GST_DEBUG_OBJECT (filter, "track tags not complete yet");
return;
}
if (album_processing && !(filter->has_album_gain && filter->has_album_peak)) {
GST_INFO_OBJECT (filter, "Album tags not complete yet");
GST_DEBUG_OBJECT (filter, "album tags not complete yet");
return;
}
if (filter->forced) {
GST_INFO_OBJECT (filter,
"Existing tags are sufficient, but processing anyway (forced)");
GST_DEBUG_OBJECT (filter,
"existing tags are sufficient, but processing anyway (forced)");
return;
}
filter->skip = TRUE;
rg_analysis_reset (filter->ctx);
if (!album_processing)
GST_INFO_OBJECT (filter,
"Existing tags are sufficient, will not process this track");
else
GST_INFO_OBJECT (filter,
"Existing tags are sufficient, will not process this album");
if (!album_processing) {
GST_DEBUG_OBJECT (filter,
"existing tags are sufficient, will not process this track");
} else {
GST_DEBUG_OBJECT (filter,
"existing tags are sufficient, will not process this album");
}
}
static void
@ -599,7 +609,9 @@ gst_rg_analysis_handle_eos (GstRgAnalysis * filter)
rg_analysis_reset_album (filter->ctx);
if (track_success || album_success) {
GST_DEBUG_OBJECT (filter, "Posting tag list with results");
GST_LOG_OBJECT (filter, "posting tag list with results");
gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
GST_TAG_REFERENCE_LEVEL, filter->reference_level, NULL);
/* This steals our reference to the list: */
gst_element_found_tags_for_pad (GST_ELEMENT (filter),
GST_BASE_TRANSFORM_SRC_PAD (GST_BASE_TRANSFORM (filter)), tag_list);
@ -609,11 +621,12 @@ gst_rg_analysis_handle_eos (GstRgAnalysis * filter)
if (album_processing) {
filter->num_tracks--;
if (!album_finished)
GST_INFO_OBJECT (filter, "Album not finished yet (num-tracks is now %u)",
if (!album_finished) {
GST_DEBUG_OBJECT (filter, "album not finished yet (num-tracks is now %u)",
filter->num_tracks);
else
GST_INFO_OBJECT (filter, "Album finished (num-tracks is now 0)");
} else {
GST_DEBUG_OBJECT (filter, "album finished (num-tracks is now 0)");
}
}
if (album_processing)
@ -631,10 +644,10 @@ gst_rg_analysis_track_result (GstRgAnalysis * filter, GstTagList ** tag_list)
if (track_success) {
track_gain += filter->reference_level - RG_REFERENCE_LEVEL;
GST_INFO_OBJECT (filter, "Track gain is %+.2f dB, peak %.6f", track_gain,
GST_INFO_OBJECT (filter, "track gain is %+.2f dB, peak %.6f", track_gain,
track_peak);
} else {
GST_INFO_OBJECT (filter, "Track was too short to analyze");
GST_INFO_OBJECT (filter, "track was too short to analyze");
}
if (track_success) {
@ -658,10 +671,10 @@ gst_rg_analysis_album_result (GstRgAnalysis * filter, GstTagList ** tag_list)
if (album_success) {
album_gain += filter->reference_level - RG_REFERENCE_LEVEL;
GST_INFO_OBJECT (filter, "Album gain is %+.2f dB, peak %.6f", album_gain,
GST_INFO_OBJECT (filter, "album gain is %+.2f dB, peak %.6f", album_gain,
album_peak);
} else {
GST_INFO_OBJECT (filter, "Album was too short to analyze");
GST_INFO_OBJECT (filter, "album was too short to analyze");
}
if (album_success) {
@ -673,14 +686,3 @@ gst_rg_analysis_album_result (GstRgAnalysis * filter, GstTagList ** tag_list)
return album_success;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rganalysis", GST_RANK_NONE,
GST_TYPE_RG_ANALYSIS);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "replaygain",
"ReplayGain analysis", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN);

2
gst/replaygain/gstrganalysis.h
View file

@ -78,6 +78,8 @@ struct _GstRgAnalysisClass
GstBaseTransformClass parent_class;
};
GType gst_rg_analysis_get_type (void);
G_END_DECLS
#endif /* __GST_RG_ANALYSIS_H__ */

197
gst/replaygain/gstrglimiter.c Normal file
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@ -0,0 +1,197 @@
/* GStreamer ReplayGain limiter
*
* Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
*
* gstrglimiter.c: Element to apply signal compression to raw audio data
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
/**
* SECTION:element-rglimiter
* @see_also: <link linkend="GstRgVolume">rgvolume</link>
*
* <refsect2>
* <para>
* This element applies signal compression/limiting to raw audio data. It
* performs strict hard limiting with soft-knee characteristics, using a
* threshold of -6 dB. This type of filter is mentioned in the proposed <ulink
* url="http://replaygain.org">ReplayGain standard</ulink>.
* </para>
* <title>Example launch line</title>
* <para>Playback of a file:</para>
* <programlisting>
* gst-launch filesrc location="Filename.ext" ! decodebin ! audioconvert \
* ! rgvolume pre-amp=6.0 headroom=10.0 ! rglimiter \
* ! audioconvert ! audioresample ! alsasink
* </programlisting>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <gst/gst.h>
#include <math.h>
#include "gstrglimiter.h"
GST_DEBUG_CATEGORY_STATIC (gst_rg_limiter_debug);
#define GST_CAT_DEFAULT gst_rg_limiter_debug
enum
{
PROP_0,
PROP_ENABLED,
};
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
"width = (int) 32, channels = (int) [1, MAX], "
"rate = (int) [1, MAX], endianness = (int) BYTE_ORDER"));
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
"width = (int) 32, channels = (int) [1, MAX], "
"rate = (int) [1, MAX], endianness = (int) BYTE_ORDER"));
GST_BOILERPLATE (GstRgLimiter, gst_rg_limiter, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM);
static void gst_rg_limiter_class_init (GstRgLimiterClass * klass);
static void gst_rg_limiter_init (GstRgLimiter * filter,
GstRgLimiterClass * gclass);
static void gst_rg_limiter_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rg_limiter_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static GstFlowReturn gst_rg_limiter_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static const GstElementDetails element_details = {
"ReplayGain limiter",
"Filter/Effect/Audio",
"Apply signal compression to raw audio data",
"Ren\xc3\xa9 Stadler <mail@renestadler.de>"
};
static void
gst_rg_limiter_base_init (gpointer g_class)
{
GstElementClass *element_class = g_class;
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_set_details (element_class, &element_details);
GST_DEBUG_CATEGORY_INIT (gst_rg_limiter_debug, "rglimiter", 0,
"ReplayGain limiter element");
}
static void
gst_rg_limiter_class_init (GstRgLimiterClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
gobject_class = (GObjectClass *) klass;
gobject_class->set_property = gst_rg_limiter_set_property;
gobject_class->get_property = gst_rg_limiter_get_property;
g_object_class_install_property (gobject_class, PROP_ENABLED,
g_param_spec_boolean ("enabled", "Enabled", "Enable processing", TRUE,
G_PARAM_READWRITE));
trans_class = GST_BASE_TRANSFORM_CLASS (klass);
trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_limiter_transform_ip);
trans_class->passthrough_on_same_caps = FALSE;
}
static void
gst_rg_limiter_init (GstRgLimiter * filter, GstRgLimiterClass * gclass)
{
filter->enabled = TRUE;
gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter), FALSE);
}
static void
gst_rg_limiter_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRgLimiter *filter = GST_RG_LIMITER (object);
switch (prop_id) {
case PROP_ENABLED:
filter->enabled = g_value_get_boolean (value);
gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
!filter->enabled);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rg_limiter_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRgLimiter *filter = GST_RG_LIMITER (object);
switch (prop_id) {
case PROP_ENABLED:
g_value_set_boolean (value, filter->enabled);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
#define LIMIT 1.0
#define THRES 0.5 /* ca. -6 dB */
#define COMPL 0.5 /* LIMIT - THRESH */
static GstFlowReturn
gst_rg_limiter_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstRgLimiter *filter = GST_RG_LIMITER (base);
gfloat *input;
guint count;
guint i;
if (!filter->enabled)
return GST_FLOW_OK;
input = (gfloat *) GST_BUFFER_DATA (buf);
count = GST_BUFFER_SIZE (buf) / sizeof (gfloat);
for (i = count; i--;) {
if (*input > THRES)
*input = tanhf ((*input - THRES) / COMPL) * COMPL + THRES;
else if (*input < -THRES)
*input = tanhf ((*input + THRES) / COMPL) * COMPL - THRES;
input++;
}
return GST_FLOW_OK;
}

64
gst/replaygain/gstrglimiter.h Normal file
View file

@ -0,0 +1,64 @@
/* GStreamer ReplayGain limiter
*
* Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
*
* gstrglimiter.h: Element to apply signal compression to raw audio data
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#ifndef __GST_RG_LIMITER_H__
#define __GST_RG_LIMITER_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#define GST_TYPE_RG_LIMITER \
(gst_rg_limiter_get_type())
#define GST_RG_LIMITER(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_LIMITER,GstRgLimiter))
#define GST_RG_LIMITER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_LIMITER,GstRgLimiterClass))
#define GST_IS_RG_LIMITER(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_LIMITER))
#define GST_IS_RG_LIMITER_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_LIMITER))
typedef struct _GstRgLimiter GstRgLimiter;
typedef struct _GstRgLimiterClass GstRgLimiterClass;
/**
* GstRgLimiter:
*
* Opaque data structure.
*/
struct _GstRgLimiter
{
GstBaseTransform element;
/*< private >*/
gboolean enabled;
};
struct _GstRgLimiterClass
{
GstBaseTransformClass parent_class;
};
GType gst_rg_limiter_get_type (void);
#endif /* __GST_RG_LIMITER_H__ */

702
gst/replaygain/gstrgvolume.c Normal file
View file

@ -0,0 +1,702 @@
/* GStreamer ReplayGain volume adjustment
*
* Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
*
* gstrgvolume.c: Element to apply ReplayGain volume adjustment
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
/**
* SECTION:element-rgvolume
* @see_also: <link linkend="GstRgLimiter">rglimiter</link>,
* <link linkend="GstRgAnalysis">rganalysis</link>
*
* <refsect2>
* <para>
* This element applies volume changes to streams as lined out in the proposed
* <ulink url="http://replaygain.org">ReplayGain standard</ulink>. It
* interprets the ReplayGain meta data tags and carries out the adjustment (by
* using a volume element internally). The relevant tags are:
* <itemizedlist>
* <listitem>#GST_TAG_TRACK_GAIN</listitem>
* <listitem>#GST_TAG_TRACK_PEAK</listitem>
* <listitem>#GST_TAG_ALBUM_GAIN</listitem>
* <listitem>#GST_TAG_ALBUM_PEAK</listitem>
* <listitem>#GST_TAG_REFERENCE_LEVEL</listitem>
* </itemizedlist>
* The information carried by these tags must have been calculated beforehand by
* performing the ReplayGain analysis. This is implemented by the <link
* linkend="GstRgAnalysis">rganalysis</link> element.
* </para>
* <para>
* The signal compression/limiting recommendations outlined in the proposed
* standard are not implemented by this element. This has to be handled by
* separate elements because applications might want to have additional filters
* between the volume adjustment and the limiting stage. A basic limiter is
* included with this plugin: The <link linkend="GstRgLimiter">rglimiter</link>
* element applies -6 dB hard limiting as mentioned in the ReplayGain standard.
* </para>
* <title>Example launch line</title>
* <para>Playback of a file:</para>
* <programlisting>
* gst-launch filesrc location="Filename.ext" ! decodebin ! audioconvert \
* ! rgvolume ! audioconvert ! audioresample ! alsasink
* </programlisting>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <gst/gst.h>
#include <gst/pbutils/pbutils.h>
#include <math.h>
#include "gstrgvolume.h"
#include "replaygain.h"
GST_DEBUG_CATEGORY_STATIC (gst_rg_volume_debug);
#define GST_CAT_DEFAULT gst_rg_volume_debug
enum
{
PROP_0,
PROP_ALBUM_MODE,
PROP_HEADROOM,
PROP_PRE_AMP,
PROP_FALLBACK_GAIN,
PROP_TARGET_GAIN,
PROP_RESULT_GAIN
};
#define DEFAULT_ALBUM_MODE TRUE
#define DEFAULT_HEADROOM 0.0
#define DEFAULT_PRE_AMP 0.0
#define DEFAULT_FALLBACK_GAIN 0.0
#define DB_TO_LINEAR(x) pow (10., (x) / 20.)
#define LINEAR_TO_DB(x) (20. * log10 (x))
#define GAIN_FORMAT "+.02f dB"
#define PEAK_FORMAT ".06f"
#define VALID_GAIN(x) ((x) > -60.00 && (x) < 60.00)
#define VALID_PEAK(x) ((x) > 0.)
/* Same template caps as GstVolume, for I don't like having just ANY caps. */
static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32; "
"audio/x-raw-int, "
"channels = (int) [ 1, MAX ], "
"rate = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE"));
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
"rate = (int) [ 1, MAX ], "
"channels = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 32; "
"audio/x-raw-int, "
"channels = (int) [ 1, MAX ], "
"rate = (int) [ 1, MAX ], "
"endianness = (int) BYTE_ORDER, "
"width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE"));
GST_BOILERPLATE (GstRgVolume, gst_rg_volume, GstBin, GST_TYPE_BIN);
static void gst_rg_volume_class_init (GstRgVolumeClass * klass);
static void gst_rg_volume_init (GstRgVolume * self, GstRgVolumeClass * gclass);
static void gst_rg_volume_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rg_volume_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static void gst_rg_volume_dispose (GObject * object);
static GstStateChangeReturn gst_rg_volume_change_state (GstElement * element,
GstStateChange transition);
static gboolean gst_rg_volume_sink_event (GstPad * pad, GstEvent * event);
static GstEvent *gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event);
static void gst_rg_volume_reset (GstRgVolume * self);
static void gst_rg_volume_update_gain (GstRgVolume * self);
static inline void gst_rg_volume_determine_gain (GstRgVolume * self,
gdouble * target_gain, gdouble * result_gain);
static void
gst_rg_volume_base_init (gpointer g_class)
{
GstElementClass *element_class = g_class;
static const GstElementDetails element_details = {
"ReplayGain volume",
"Filter/Effect/Audio",
"Apply ReplayGain volume adjustment",
"Ren\xc3\xa9 Stadler <mail@renestadler.de>"
};
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_template));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_template));
gst_element_class_set_details (element_class, &element_details);
GST_DEBUG_CATEGORY_INIT (gst_rg_volume_debug, "rgvolume", 0,
"ReplayGain volume element");
}
static void
gst_rg_volume_class_init (GstRgVolumeClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *element_class;
GstBinClass *bin_class;
gobject_class = (GObjectClass *) klass;
gobject_class->set_property = gst_rg_volume_set_property;
gobject_class->get_property = gst_rg_volume_get_property;
gobject_class->dispose = gst_rg_volume_dispose;
/**
* GstRgVolume:album-mode:
*
* Whether to prefer album gain over track gain.
*
* If set to %TRUE, use album gain instead of track gain if both are
* available. This keeps the relative loudness levels of tracks from the same
* album intact.
*
* If set to %FALSE, track mode is used instead. This effectively leads to
* more extensive normalization.
*
* If album mode is enabled but the album gain tag is absent in the stream,
* the track gain is used instead. If both gain tags are missing, the value
* of the <link linkend="GstRgVolume--fallback-gain">fallback-gain</link>
* property is used instead.
*/
g_object_class_install_property (gobject_class, PROP_ALBUM_MODE,
g_param_spec_boolean ("album-mode", "Album mode",
"Prefer album over track gain", DEFAULT_ALBUM_MODE,
G_PARAM_READWRITE));
/**
* GstRgVolume:headroom:
*
* Extra headroom [dB]. This controls the amount by which the output can
* exceed digital full scale.
*
* Only set this to a value greater than 0.0 if signal compression/limiting of
* a suitable form is applied to the output (or output is brought into the
* correct range by some other transformation).
*
* This element internally uses a volume element, which also supports
* operating on integer audio formats. These formats do not allow exceeding
* digital full scale. If extra headroom is used, make sure that the raw
* audio data format is floating point (audio/x-raw-float). Otherwise,
* clipping distortion might be introduced as part of the volume adjustment
* itself.
*/
g_object_class_install_property (gobject_class, PROP_HEADROOM,
g_param_spec_double ("headroom", "Headroom", "Extra headroom [dB]",
0., 60., DEFAULT_HEADROOM, G_PARAM_READWRITE));
/**
* GstRgVolume:pre-amp:
*
* Additional gain to apply globally [dB]. This controls the trade-off
* between uniformity of normalization and utilization of available dynamic
* range.
*
* Note that the default value is 0 dB because the ReplayGain reference value
* was adjusted by +6 dB (from 83 to 89 dB). At the time of this writing, the
* <ulink url="http://replaygain.org">webpage</ulink> is still outdated and
* does not reflect this change however. Where the original proposal states
* that a proper default pre-amp value is +6 dB, this translates to the used 0
* dB.
*/
g_object_class_install_property (gobject_class, PROP_PRE_AMP,
g_param_spec_double ("pre-amp", "Pre-amp", "Extra gain [dB]",
-60., 60., DEFAULT_PRE_AMP, G_PARAM_READWRITE));
/**
* GstRgVolume:fallback-gain:
*
* Fallback gain [dB] for streams missing ReplayGain tags.
*/
g_object_class_install_property (gobject_class, PROP_FALLBACK_GAIN,
g_param_spec_double ("fallback-gain", "Fallback gain",
"Gain for streams missing tags [dB]",
-60., 60., DEFAULT_FALLBACK_GAIN, G_PARAM_READWRITE));
/**
* GstRgVolume:result-gain:
*
* Applied gain [dB]. This gain is applied to processed buffer data.
*
* This is set to the <link linkend="GstRgVolume--target-gain">target
* gain</link> if amplification by that amount can be applied safely.
* "Safely" means that the volume adjustment does not inflict clipping
* distortion. Should this not be the case, the result gain is set to an
* appropriately reduced value (by applying peak normalization). The proposed
* standard calls this "clipping prevention".
*
* The difference between target and result gain reflects the necessary amount
* of reduction. Applications can make use of this information to temporarily
* reduce the <link linkend="GstRgVolume--pre-amp">pre-amp</link> for
* subsequent streams, as recommended by the ReplayGain standard.
*
* Note that target and result gain differing for a great majority of streams
* indicates a problem: What happens in this case is that most streams receive
* peak normalization instead of amplification by the ideal replay gain. To
* prevent this, the <link linkend="GstRgVolume--pre-amp">pre-amp</link> has
* to be lowered and/or a limiter has to be used which facilitates the use of
* <link linkend="GstRgVolume--headroom">headroom</link>.
*/
g_object_class_install_property (gobject_class, PROP_RESULT_GAIN,
g_param_spec_double ("result-gain", "Result-gain", "Applied gain [dB]",
-120., 120., 0., G_PARAM_READABLE));
/**
* GstRgVolume:target-gain:
*
* Applicable gain [dB]. This gain is supposed to be applied.
*
* Depending on the value of the <link
* linkend="GstRgVolume--album-mode">album-mode</link> property and the
* presence of ReplayGain tags in the stream, this is set according to one of
* these simple formulas:
*
* <itemizedlist>
* <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + album gain
* of the stream</listitem>
* <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + track gain
* of the stream</listitem>
* <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + <link
* linkend="GstRgVolume--fallback-gain">fallback gain</link></listitem>
* </itemizedlist>
*/
g_object_class_install_property (gobject_class, PROP_TARGET_GAIN,
g_param_spec_double ("target-gain", "Target-gain",
"Applicable gain [dB]", -120., 120., 0., G_PARAM_READABLE));
element_class = (GstElementClass *) klass;
element_class->change_state = GST_DEBUG_FUNCPTR (gst_rg_volume_change_state);
bin_class = (GstBinClass *) klass;
/* Setting these to NULL makes gst_bin_add and _remove refuse to let anyone
* mess with our internals. */
bin_class->add_element = NULL;
bin_class->remove_element = NULL;
}
static void
gst_rg_volume_init (GstRgVolume * self, GstRgVolumeClass * gclass)
{
GObjectClass *volume_class;
GstPad *volume_pad, *ghost_pad;
self->album_mode = DEFAULT_ALBUM_MODE;
self->headroom = DEFAULT_HEADROOM;
self->pre_amp = DEFAULT_PRE_AMP;
self->fallback_gain = DEFAULT_FALLBACK_GAIN;
self->target_gain = 0.0;
self->result_gain = 0.0;
self->volume_element = gst_element_factory_make ("volume", "rgvolume-volume");
if (G_UNLIKELY (self->volume_element == NULL)) {
GstMessage *msg;
GST_WARNING_OBJECT (self, "could not create volume element");
msg = gst_missing_element_message_new (GST_ELEMENT_CAST (self), "volume");
gst_element_post_message (GST_ELEMENT_CAST (self), msg);
/* Nothing else to do, we will refuse the state change from NULL to READY to
* indicate that something went very wrong. It is doubtful that someone
* attempts changing our state though, since we end up having no pads! */
return;
}
volume_class = G_OBJECT_GET_CLASS (G_OBJECT (self->volume_element));
self->max_volume = G_PARAM_SPEC_DOUBLE
(g_object_class_find_property (volume_class, "volume"))->maximum;
GST_BIN_CLASS (parent_class)->add_element (GST_BIN_CAST (self),
self->volume_element);
volume_pad = gst_element_get_pad (self->volume_element, "sink");
ghost_pad = gst_ghost_pad_new_from_template ("sink", volume_pad,
gst_pad_get_pad_template (volume_pad));
gst_object_unref (volume_pad);
gst_pad_set_event_function (ghost_pad, gst_rg_volume_sink_event);
gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad);
volume_pad = gst_element_get_pad (self->volume_element, "src");
ghost_pad = gst_ghost_pad_new_from_template ("src", volume_pad,
gst_pad_get_pad_template (volume_pad));
gst_object_unref (volume_pad);
gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad);
}
static void
gst_rg_volume_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRgVolume *self = GST_RG_VOLUME (object);
switch (prop_id) {
case PROP_ALBUM_MODE:
self->album_mode = g_value_get_boolean (value);
break;
case PROP_HEADROOM:
self->headroom = g_value_get_double (value);
break;
case PROP_PRE_AMP:
self->pre_amp = g_value_get_double (value);
break;
case PROP_FALLBACK_GAIN:
self->fallback_gain = g_value_get_double (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
gst_rg_volume_update_gain (self);
}
static void
gst_rg_volume_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRgVolume *self = GST_RG_VOLUME (object);
switch (prop_id) {
case PROP_ALBUM_MODE:
g_value_set_boolean (value, self->album_mode);
break;
case PROP_HEADROOM:
g_value_set_double (value, self->headroom);
break;
case PROP_PRE_AMP:
g_value_set_double (value, self->pre_amp);
break;
case PROP_FALLBACK_GAIN:
g_value_set_double (value, self->fallback_gain);
break;
case PROP_TARGET_GAIN:
g_value_set_double (value, self->target_gain);
break;
case PROP_RESULT_GAIN:
g_value_set_double (value, self->result_gain);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rg_volume_dispose (GObject * object)
{
GstRgVolume *self = GST_RG_VOLUME (object);
if (self->volume_element != NULL) {
/* Manually remove our child using the bin implementation of remove_element.
* This is needed because we prevent gst_bin_remove from working, which the
* parent dispose handler would use if we had any children left. */
GST_BIN_CLASS (parent_class)->remove_element (GST_BIN_CAST (self),
self->volume_element);
self->volume_element = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static GstStateChangeReturn
gst_rg_volume_change_state (GstElement * element, GstStateChange transition)
{
GstRgVolume *self = GST_RG_VOLUME (element);
GstStateChangeReturn res;
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
if (G_UNLIKELY (self->volume_element == NULL)) {
/* Creating our child volume element in _init failed. */
return GST_STATE_CHANGE_FAILURE;
}
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_rg_volume_reset (self);
break;
default:
break;
}
res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
return res;
}
/* Event function for the ghost sink pad. */
static gboolean
gst_rg_volume_sink_event (GstPad * pad, GstEvent * event)
{
GstRgVolume *self;
GstPad *volume_sink_pad;
GstEvent *send_event = event;
gboolean res;
self = GST_RG_VOLUME (gst_pad_get_parent_element (pad));
volume_sink_pad = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_TAG:
GST_LOG_OBJECT (self, "received tag event");
send_event = gst_rg_volume_tag_event (self, event);
if (send_event == NULL)
GST_LOG_OBJECT (self, "all tags handled, dropping event");
break;
case GST_EVENT_EOS:
gst_rg_volume_reset (self);
break;
default:
break;
}
if (G_LIKELY (send_event != NULL))
res = gst_pad_send_event (volume_sink_pad, send_event);
else
res = TRUE;
gst_object_unref (volume_sink_pad);
gst_object_unref (self);
return res;
}
static GstEvent *
gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event)
{
GstTagList *tag_list;
gboolean has_track_gain, has_track_peak, has_album_gain, has_album_peak;
gboolean has_ref_level;
g_return_val_if_fail (event != NULL, NULL);
g_return_val_if_fail (GST_EVENT_TYPE (event) == GST_EVENT_TAG, event);
gst_event_parse_tag (event, &tag_list);
if (gst_tag_list_is_empty (tag_list))
return event;
has_track_gain = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN,
&self->track_gain);
has_track_peak = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK,
&self->track_peak);
has_album_gain = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN,
&self->album_gain);
has_album_peak = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK,
&self->album_peak);
has_ref_level = gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
&self->reference_level);
if (!has_track_gain && !has_track_peak && !has_album_gain && !has_album_peak)
return event;
if (has_ref_level && (has_track_gain || has_album_gain)
&& (ABS (self->reference_level - RG_REFERENCE_LEVEL) > 1.e-6)) {
/* Log a message stating the amount of adjustment that is applied below. */
GST_DEBUG_OBJECT (self,
"compensating for reference level difference by %" GAIN_FORMAT,
RG_REFERENCE_LEVEL - self->reference_level);
}
if (has_track_gain) {
self->track_gain += RG_REFERENCE_LEVEL - self->reference_level;
}
if (has_album_gain) {
self->album_gain += RG_REFERENCE_LEVEL - self->reference_level;
}
/* Ignore values that are obviously invalid. */
if (G_UNLIKELY (has_track_gain && !VALID_GAIN (self->track_gain))) {
GST_DEBUG_OBJECT (self,
"ignoring bogus track gain value %" GAIN_FORMAT, self->track_gain);
has_track_gain = FALSE;
}
if (G_UNLIKELY (has_track_peak && !VALID_PEAK (self->track_peak))) {
GST_DEBUG_OBJECT (self,
"ignoring bogus track peak value %" PEAK_FORMAT, self->track_peak);
has_track_peak = FALSE;
}
if (G_UNLIKELY (has_album_gain && !VALID_GAIN (self->album_gain))) {
GST_DEBUG_OBJECT (self,
"ignoring bogus album gain value %" GAIN_FORMAT, self->album_gain);
has_album_gain = FALSE;
}
if (G_UNLIKELY (has_album_peak && !VALID_PEAK (self->album_peak))) {
GST_DEBUG_OBJECT (self,
"ignoring bogus album peak value %" PEAK_FORMAT, self->album_peak);
has_album_peak = FALSE;
}
self->has_track_gain |= has_track_gain;
self->has_track_peak |= has_track_peak;
self->has_album_gain |= has_album_gain;
self->has_album_peak |= has_album_peak;
event = (GstEvent *) gst_mini_object_make_writable (GST_MINI_OBJECT (event));
gst_event_parse_tag (event, &tag_list);
gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_GAIN);
gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_PEAK);
gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_GAIN);
gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_PEAK);
gst_tag_list_remove_tag (tag_list, GST_TAG_REFERENCE_LEVEL);
gst_rg_volume_update_gain (self);
if (gst_tag_list_is_empty (tag_list)) {
gst_event_unref (event);
event = NULL;
}
return event;
}
static void
gst_rg_volume_reset (GstRgVolume * self)
{
self->has_track_gain = FALSE;
self->has_track_peak = FALSE;
self->has_album_gain = FALSE;
self->has_album_peak = FALSE;
self->reference_level = RG_REFERENCE_LEVEL;
gst_rg_volume_update_gain (self);
}
static void
gst_rg_volume_update_gain (GstRgVolume * self)
{
gdouble target_gain, result_gain, result_volume;
gboolean target_gain_changed, result_gain_changed;
gst_rg_volume_determine_gain (self, &target_gain, &result_gain);
result_volume = DB_TO_LINEAR (result_gain);
/* Ensure that the result volume is within the range that the volume element
* can handle. Currently, the limit is 10. (+20 dB), which should not be
* restrictive. */
if (G_UNLIKELY (result_volume > self->max_volume)) {
GST_INFO_OBJECT (self,
"cannot handle result gain of %" GAIN_FORMAT " (%0.6f), adjusting",
result_gain, result_volume);
result_volume = self->max_volume;
result_gain = LINEAR_TO_DB (result_volume);
}
/* Direct comparison is OK in this case. */
if (target_gain == result_gain) {
GST_INFO_OBJECT (self,
"result gain is %" GAIN_FORMAT " (%0.6f), matching target",
result_gain, result_volume);
} else {
GST_INFO_OBJECT (self,
"result gain is %" GAIN_FORMAT " (%0.6f), target is %" GAIN_FORMAT,
result_gain, result_volume, target_gain);
}
target_gain_changed = (self->target_gain != target_gain);
result_gain_changed = (self->result_gain != result_gain);
self->target_gain = target_gain;
self->result_gain = result_gain;
g_object_set (self->volume_element, "volume", result_volume, NULL);
if (target_gain_changed)
g_object_notify ((GObject *) self, "target-gain");
if (result_gain_changed)
g_object_notify ((GObject *) self, "result-gain");
}
static inline void
gst_rg_volume_determine_gain (GstRgVolume * self, gdouble * target_gain,
gdouble * result_gain)
{
gdouble gain, peak;
if (!self->has_track_gain && !self->has_album_gain) {
GST_DEBUG_OBJECT (self, "using fallback gain");
gain = self->fallback_gain;
peak = 1.0;
} else if ((self->album_mode && self->has_album_gain)
|| (!self->album_mode && !self->has_track_gain)) {
gain = self->album_gain;
if (G_LIKELY (self->has_album_peak)) {
peak = self->album_peak;
} else {
GST_DEBUG_OBJECT (self, "album peak missing, assuming 1.0");
peak = 1.0;
}
/* Falling back from track to album gain shouldn't really happen. */
if (G_UNLIKELY (!self->album_mode))
GST_INFO_OBJECT (self, "falling back to album gain");
} else {
/* !album_mode && !has_album_gain || album_mode && has_track_gain */
gain = self->track_gain;
if (G_LIKELY (self->has_track_peak)) {
peak = self->track_peak;
} else {
GST_DEBUG_OBJECT (self, "track peak missing, assuming 1.0");
peak = 1.0;
}
if (self->album_mode)
GST_INFO_OBJECT (self, "falling back to track gain");
}
gain += self->pre_amp;
*target_gain = gain;
*result_gain = gain;
if (LINEAR_TO_DB (peak) + gain > self->headroom) {
*result_gain = LINEAR_TO_DB (1. / peak) + self->headroom;
}
}

88
gst/replaygain/gstrgvolume.h Normal file
View file

@ -0,0 +1,88 @@
/* GStreamer ReplayGain volume adjustment
*
* Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
*
* gstrgvolume.h: Element to apply ReplayGain volume adjustment
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#ifndef __GST_RG_VOLUME_H__
#define __GST_RG_VOLUME_H__
#include <gst/gst.h>
G_BEGIN_DECLS
#define GST_TYPE_RG_VOLUME \
(gst_rg_volume_get_type())
#define GST_RG_VOLUME(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_VOLUME,GstRgVolume))
#define GST_RG_VOLUME_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_VOLUME,GstRgVolumeClass))
#define GST_IS_PLUGIN_TEMPLATE(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_VOLUME))
#define GST_IS_PLUGIN_TEMPLATE_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_VOLUME))
typedef struct _GstRgVolume GstRgVolume;
typedef struct _GstRgVolumeClass GstRgVolumeClass;
/**
* GstRgVolume:
*
* Opaque data structure.
*/
struct _GstRgVolume
{
GstBin bin;
/*< private >*/
GstElement *volume_element;
gdouble max_volume;
gboolean album_mode;
gdouble headroom;
gdouble pre_amp;
gdouble fallback_gain;
gdouble target_gain;
gdouble result_gain;
gdouble track_gain;
gdouble track_peak;
gdouble album_gain;
gdouble album_peak;
gboolean has_track_gain;
gboolean has_track_peak;
gboolean has_album_gain;
gboolean has_album_peak;
gdouble reference_level;
};
struct _GstRgVolumeClass
{
GstBinClass parent_class;
};
GType gst_rg_volume_get_type (void);
G_END_DECLS
#endif /* __GST_RG_VOLUME_H__ */

53
gst/replaygain/replaygain.c Normal file
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@ -0,0 +1,53 @@
/* GStreamer ReplayGain plugin
*
* Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
*
* replaygain.c: Plugin providing ReplayGain related elements
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <gst/gst.h>
#include "gstrganalysis.h"
#include "gstrglimiter.h"
#include "gstrgvolume.h"
static gboolean
plugin_init (GstPlugin * plugin)
{
if (!gst_element_register (plugin, "rganalysis", GST_RANK_NONE,
GST_TYPE_RG_ANALYSIS))
return FALSE;
if (!gst_element_register (plugin, "rglimiter", GST_RANK_NONE,
GST_TYPE_RG_LIMITER))
return FALSE;
if (!gst_element_register (plugin, "rgvolume", GST_RANK_NONE,
GST_TYPE_RG_VOLUME))
return FALSE;
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "replaygain",
"ReplayGain volume normalization", plugin_init, VERSION, GST_LICENSE,
GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);

36
gst/replaygain/replaygain.h Normal file
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@ -0,0 +1,36 @@
/* GStreamer ReplayGain plugin
*
* Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
*
* replaygain.h: Plugin providing ReplayGain related elements
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#ifndef __REPLAYGAIN_H__
#define __REPLAYGAIN_H__
G_BEGIN_DECLS
/* Reference level (in dBSPL). The 2001 proposal specifies 83. This was
* changed later in all implementations to 89, which is the new, offical value:
* David Robinson acknowledged the change but didn't update the website yet. */
#define RG_REFERENCE_LEVEL 89.
G_END_DECLS
#endif /* __REPLAYGAIN_H__ */

2
gst/replaygain/rganalysis.h
View file

@ -29,8 +29,6 @@
G_BEGIN_DECLS
#define RG_REFERENCE_LEVEL 89.
typedef struct _RgAnalysisCtx RgAnalysisCtx;
RgAnalysisCtx *rg_analysis_new (void);

6
tests/check/Makefile.am
View file

@ -57,9 +57,12 @@ VALGRIND_TESTS_DISABLE = \
$(VALGRIND_TO_FIX)
check_PROGRAMS = \
elements/deinterleave \
$(check_mpeg2enc) \
$(check_neon) \
elements/rganalysis \
elements/rglimiter \
elements/rgvolume \
elements/videocrop \
$(check_wavpack) \
elements/y4menc
@ -72,3 +75,6 @@ LDADD = $(GST_OBJ_LIBS) $(GST_CHECK_LIBS) $(CHECK_LIBS)
elements_videocrop_LDADD = $(LDADD) $(GST_BASE_LIBS)
elements_videocrop_CFLAGS = $(CFLAGS) $(AM_CFLAGS) $(GST_BASE_CFLAGS)
elements_deinterleave_LDADD = $(LDADD) $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR)
elements_deinterleave_CFLAGS = $(CFLAGS) $(AM_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)

4
tests/check/elements/.gitignore vendored
View file

@ -1,8 +1,10 @@
.dirstamp
tagid3v2mux
deinterleave
gdpdepay
gdppay
mpeg2enc
rglimiter
rgvolume
wavpackdec
wavpackenc
wavpackparse

187
tests/check/elements/rganalysis.c
View file

@ -20,77 +20,72 @@
* 02110-1301 USA
*/
/* Some things to note about the RMS window length of the analysis
* algorithm and thus the implementation used in the element:
* Processing divides input data into 50ms windows at some point.
* Some details about this that normally do not matter:
/* Some things to note about the RMS window length of the analysis algorithm and
* thus the implementation used in the element: Processing divides input data
* into 50ms windows at some point. Some details about this that normally do
* not matter:
*
* 1. At the end of a stream, the remainder of data that did not fill
* up the last 50ms window is simply discarded.
* 1. At the end of a stream, the remainder of data that did not fill up the
* last 50ms window is simply discarded.
*
* 2. If the sample rate changes during a stream, the currently
* running window is discarded and the equal loudness filter gets
* reset as if a new stream started.
* 2. If the sample rate changes during a stream, the currently running window
* is discarded and the equal loudness filter gets reset as if a new stream
* started.
*
* 3. For the album gain, it is not entirely correct to think of
* obtaining it like "as if all the tracks are analyzed as one
* track". There isn't a separate window being tracked for album
* processing, so at stream (track) end, the remaining unfilled
* window does not contribute to the album gain either.
* 3. For the album gain, it is not entirely correct to think of obtaining it
* like "as if all the tracks are analyzed as one track". There isn't a
* separate window being tracked for album processing, so at stream (track)
* end, the remaining unfilled window does not contribute to the album gain
* either.
*
* 4. If a waveform with a result gain G is concatenated to itself
* and the result processed as a track, the gain can be different
* from G if and only if the duration of the original waveform is
* not an integer multiple of 50ms. If the original waveform gets
* processed as a single track and then the same data again as a
* subsequent track, the album result gain will always match G
* (this is implied by 3.).
* 4. If a waveform with a result gain G is concatenated to itself and the
* result processed as a track, the gain can be different from G if and only
* if the duration of the original waveform is not an integer multiple of
* 50ms. If the original waveform gets processed as a single track and then
* the same data again as a subsequent track, the album result gain will
* always match G (this is implied by 3.).
*
* 5. A stream shorter than 50ms cannot be analyzed. At 8000 and
* 48000 Hz, this corresponds to 400 resp. 2400 frames. If a
* stream is shorter than 50ms, the element will not generate tags
* at EOS (only if an album finished, but only album tags are
* generated then). This is not an erroneous condition, the
* element should behave normally.
* 5. A stream shorter than 50ms cannot be analyzed. At 8000 and 48000 Hz,
* this corresponds to 400 resp. 2400 frames. If a stream is shorter than
* 50ms, the element will not generate tags at EOS (only if an album
* finished, but only album tags are generated then). This is not an
* erroneous condition, the element should behave normally.
*
* The limitations outlined in 1.-4. do not apply to the peak values.
* Every single sample is accounted for when looking for the peak.
* Thus the album peak is guaranteed to be the maximum value of all
* track peaks.
* The limitations outlined in 1.-4. do not apply to the peak values. Every
* single sample is accounted for when looking for the peak. Thus the album
* peak is guaranteed to be the maximum value of all track peaks.
*
* In normal day-to-day use, these little facts are unlikely to be
* relevant, but they have to be kept in mind for writing the tests
* here.
* In normal day-to-day use, these little facts are unlikely to be relevant, but
* they have to be kept in mind for writing the tests here.
*/
#include <gst/check/gstcheck.h>
GList *buffers = NULL;
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
/* For ease of programming we use globals to keep refs for our floating src and
* sink pads we create; otherwise we always have to do get_pad, get_peer, and
* then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
/* Mapping from supported sample rates to the correct result gain for
* the following test waveform: 20 * 512 samples with a quarter-full
* amplitude of toggling sign, changing every 48 samples and starting
* with the positive value.
/* Mapping from supported sample rates to the correct result gain for the
* following test waveform: 20 * 512 samples with a quarter-full amplitude of
* toggling sign, changing every 48 samples and starting with the positive
* value.
*
* Even if we would generate a wave describing a signal with the same
* frequency at each sampling rate, the results would vary (slightly).
* Hence the simple generation method, since we cannot use a constant
* value as expected result anyways. For all sample rates, changing
* the sign every 48 frames gives a sane frequency. Buffers
* containing data that forms such a waveform is created using the
* test_buffer_square_{float,int16}_{mono,stereo} functions below.
* Even if we would generate a wave describing a signal with the same frequency
* at each sampling rate, the results would vary (slightly). Hence the simple
* generation method, since we cannot use a constant value as expected result
* anyways. For all sample rates, changing the sign every 48 frames gives a
* sane frequency. Buffers containing data that forms such a waveform is
* created using the test_buffer_square_{float,int16}_{mono,stereo} functions
* below.
*
* The results have been checked against what the metaflac and
* wavegain programs generate for such a stream. If you want to
* verify these, be sure that the metaflac program does not produce
* incorrect results in your environment: I found a strange bug in the
* (defacto) reference code for the analysis that sometimes leads to
* incorrect RMS window lengths. */
* The results have been checked against what the metaflac and wavegain programs
* generate for such a stream. If you want to verify these, be sure that the
* metaflac program does not produce incorrect results in your environment: I
* found a strange bug in the (defacto) reference code for the analysis that
* sometimes leads to incorrect RMS window lengths. */
struct rate_test
{
@ -212,11 +207,10 @@ send_eos_event (GstElement * element)
fail_unless (gst_pad_send_event (pad, event),
"Cannot send EOS event: Not handled.");
/* There is no sink element, so _we_ post the EOS message on the bus
* here. Of course we generate any EOS ourselves, but this allows
* us to poll for the EOS message in poll_eos if we expect the
* element to _not_ generate a TAG message. That's better than
* waiting for a timeout to lapse. */
/* There is no sink element, so _we_ post the EOS message on the bus here. Of
* course we generate any EOS ourselves, but this allows us to poll for the
* EOS message in poll_eos if we expect the element to _not_ generate a TAG
* message. That's better than waiting for a timeout to lapse. */
fail_unless (gst_bus_post (bus, gst_message_new_eos (NULL)));
gst_object_unref (bus);
@ -251,8 +245,8 @@ poll_eos (GstElement * element)
gst_object_unref (bus);
}
/* This also polls for EOS since the TAG message comes right before
* the end of streams. */
/* This also polls for EOS since the TAG message comes right before the end of
* streams. */
static GstTagList *
poll_tags (GstElement * element)
@ -749,14 +743,13 @@ GST_END_TEST;
/* Tests for correctness of the peak values. */
/* Float peak test. For stereo, one channel has the constant value of
* -1.369, the other one 0.0. This tests many things: The result peak
* value should occur on any channel. The peak is of course the
* absolute amplitude, so 1.369 should be the result. This will also
* detect if the code uses the absolute value during the comparison.
* If it is buggy it will return 0.0 since 0.0 > -1.369. Furthermore,
* this makes sure that there is no problem with headroom (exceeding
* 0dBFS). In the wild you get float samples > 1.0 from stuff like
/* Float peak test. For stereo, one channel has the constant value of -1.369,
* the other one 0.0. This tests many things: The result peak value should
* occur on any channel. The peak is of course the absolute amplitude, so 1.369
* should be the result. This will also detect if the code uses the absolute
* value during the comparison. If it is buggy it will return 0.0 since 0.0 >
* -1.369. Furthermore, this makes sure that there is no problem with headroom
* (exceeding 0dBFS). In the wild you get float samples > 1.0 from stuff like
* vorbis. */
GST_START_TEST (test_peak_float)
@ -1089,11 +1082,10 @@ GST_START_TEST (test_peak_track_album)
GST_END_TEST;
/* Disabling album processing before the end of the album. Probably a
* rare edge case and applications should not rely on this to work.
* They need to send the element to the READY state to clear up after
* an aborted album anyway since they might need to process another
* album afterwards. */
/* Disabling album processing before the end of the album. Probably a rare edge
* case and applications should not rely on this to work. They need to send the
* element to the READY state to clear up after an aborted album anyway since
* they might need to process another album afterwards. */
GST_START_TEST (test_peak_album_abort_to_track)
{
@ -1136,8 +1128,8 @@ GST_START_TEST (test_gain_album)
g_object_set (element, "num-tracks", 3, NULL);
set_playing_state (element);
/* The three tracks are constructed such that if any of these is in
* fact ignored for the album gain, the album gain will differ. */
/* The three tracks are constructed such that if any of these is in fact
* ignored for the album gain, the album gain will differ. */
accumulator = 0;
for (i = 8; i--;)
@ -1268,12 +1260,11 @@ GST_START_TEST (test_forced_separate)
GST_END_TEST;
/* A TAG event is sent _after_ data has already been processed. In
* real pipelines, this could happen if there is more than one
* rganalysis element (by accident). While it would have analyzed all
* the data prior to receiving the event, I expect it to not post its
* results if not forced. This test is almost equivalent to
* test_forced. */
/* A TAG event is sent _after_ data has already been processed. In real
* pipelines, this could happen if there is more than one rganalysis element (by
* accident). While it would have analyzed all the data prior to receiving the
* event, I expect it to not post its results if not forced. This test is
* almost equivalent to test_forced. */
GST_START_TEST (test_forced_after_data)
{
@ -1311,8 +1302,8 @@ GST_START_TEST (test_forced_after_data)
GST_END_TEST;
/* Like test_forced, but *analyze* an album afterwards. The two tests
* following this one check the *skipping* of albums. */
/* Like test_forced, but *analyze* an album afterwards. The two tests following
* this one check the *skipping* of albums. */
GST_START_TEST (test_forced_album)
{
@ -1441,9 +1432,8 @@ GST_START_TEST (test_forced_album_no_skip)
gst_tag_list_free (tag_list);
fail_unless_num_tracks (element, 1);
/* The second track has indeed full tags, but although being not
* forced, this one has to be processed because album processing is
* on. */
/* The second track has indeed full tags, but although being not forced, this
* one has to be processed because album processing is on. */
tag_list = gst_tag_list_new ();
/* Provided values are totally arbitrary. */
gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
@ -1515,12 +1505,27 @@ GST_START_TEST (test_reference_level)
{
GstElement *element = setup_rganalysis ();
GstTagList *tag_list;
gdouble ref_level;
gint accumulator = 0;
gint i;
g_object_set (element, "reference-level", 83., "num-tracks", 2, NULL);
set_playing_state (element);
for (i = 20; i--;)
push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
0.25, 0.25));
send_eos_event (element);
tag_list = poll_tags (element);
fail_unless_track_peak (tag_list, 0.25);
fail_unless_track_gain (tag_list, get_expected_gain (44100));
fail_if_album_tags (tag_list);
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));
@ -1529,6 +1534,9 @@ GST_START_TEST (test_reference_level)
fail_unless_track_peak (tag_list, 0.25);
fail_unless_track_gain (tag_list, get_expected_gain (44100) - 6.);
fail_if_album_tags (tag_list);
fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
&ref_level) && MATCH_GAIN (ref_level, 83.),
"Incorrect reference level tag");
gst_tag_list_free (tag_list);
accumulator = 0;
@ -1543,6 +1551,9 @@ GST_START_TEST (test_reference_level)
/* We provided the same waveform twice, with a reset separating
* them. Therefore, the album gain matches the track gain. */
fail_unless_album_gain (tag_list, get_expected_gain (44100) - 6.);
fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
&ref_level) && MATCH_GAIN (ref_level, 83.),
"Incorrect reference level tag");
gst_tag_list_free (tag_list);
cleanup_rganalysis (element);

238
tests/check/elements/rglimiter.c Normal file
View file

@ -0,0 +1,238 @@
/* GStreamer ReplayGain limiter
*
* Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
*
* rglimiter.c: Unit test for the rglimiter element
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#include <gst/check/gstcheck.h>
#include <math.h>
GList *buffers = NULL;
/* For ease of programming we use globals to keep refs for our floating
* src and sink pads we create; otherwise we always have to do get_pad,
* get_peer, and then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
#define RG_LIMITER_CAPS_TEMPLATE_STRING \
"audio/x-raw-float, " \
"width = (int) 32, " \
"endianness = (int) BYTE_ORDER, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ]"
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (RG_LIMITER_CAPS_TEMPLATE_STRING)
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (RG_LIMITER_CAPS_TEMPLATE_STRING)
);
GstElement *
setup_rglimiter ()
{
GstElement *element;
GstBus *bus;
GST_DEBUG ("setup_rglimiter");
element = gst_check_setup_element ("rglimiter");
mysrcpad = gst_check_setup_src_pad (element, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (element, &sinktemplate, NULL);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return element;
}
void
cleanup_rglimiter (GstElement * element)
{
GST_DEBUG ("cleanup_rglimiter");
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
gst_check_teardown_src_pad (element);
gst_check_teardown_sink_pad (element);
gst_check_teardown_element (element);
}
static void
set_playing_state (GstElement * element)
{
fail_unless (gst_element_set_state (element,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"Could not set state to PLAYING");
}
static const gfloat test_input[] = {
-2.0, -1.0, -0.75, -0.5, -0.25, 0.0, 0.25, 0.5, 0.75, 1.0, 2.0
};
static const gfloat test_output[] = {
-0.99752737684336523, /* -2.0 */
-0.88079707797788243, /* -1.0 */
-0.7310585786300049, /* -0.75 */
-0.5, -0.25, 0.0, 0.25, 0.5,
0.7310585786300049, /* 0.75 */
0.88079707797788243, /* 1.0 */
0.99752737684336523, /* 2.0 */
};
static GstBuffer *
create_test_buffer ()
{
GstBuffer *buf = gst_buffer_new_and_alloc (sizeof (test_input));
GstCaps *caps;
memcpy (GST_BUFFER_DATA (buf), test_input, sizeof (test_input));
caps = gst_caps_new_simple ("audio/x-raw-float",
"rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1,
"endianess", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL);
gst_buffer_set_caps (buf, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
return buf;
}
static void
verify_test_buffer (GstBuffer * buf)
{
gfloat *output = (gfloat *) GST_BUFFER_DATA (buf);
gint i;
fail_unless (GST_BUFFER_SIZE (buf) == sizeof (test_output));
for (i = 0; i < G_N_ELEMENTS (test_input); i++)
fail_unless (ABS (output[i] - test_output[i]) < 1.e-6,
"Incorrect output value %.6f for input %.2f, expected %.6f",
output[i], test_input[i], test_output[i]);
}
/* Start of tests. */
GST_START_TEST (test_no_buffer)
{
GstElement *element = setup_rglimiter ();
set_playing_state (element);
cleanup_rglimiter (element);
}
GST_END_TEST;
GST_START_TEST (test_disabled)
{
GstElement *element = setup_rglimiter ();
GstBuffer *buf, *out_buf;
g_object_set (element, "enabled", FALSE, NULL);
set_playing_state (element);
buf = create_test_buffer ();
fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
fail_unless (g_list_length (buffers) == 1);
out_buf = buffers->data;
fail_if (out_buf == NULL);
buffers = g_list_remove (buffers, out_buf);
ASSERT_BUFFER_REFCOUNT (out_buf, "out_buf", 1);
fail_unless (buf == out_buf);
gst_buffer_unref (out_buf);
cleanup_rglimiter (element);
}
GST_END_TEST;
GST_START_TEST (test_limiting)
{
GstElement *element = setup_rglimiter ();
GstBuffer *buf, *out_buf;
set_playing_state (element);
/* Mutable variant. */
buf = create_test_buffer ();
fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
fail_unless (g_list_length (buffers) == 1);
out_buf = buffers->data;
fail_if (out_buf == NULL);
ASSERT_BUFFER_REFCOUNT (out_buf, "out_buf", 1);
verify_test_buffer (out_buf);
/* Immutable variant. */
buf = create_test_buffer ();
/* Extra ref: */
gst_buffer_ref (buf);
ASSERT_BUFFER_REFCOUNT (buf, "buf", 2);
fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
fail_unless (g_list_length (buffers) == 2);
out_buf = g_list_last (buffers)->data;
fail_if (out_buf == NULL);
ASSERT_BUFFER_REFCOUNT (out_buf, "out_buf", 1);
fail_unless (buf != out_buf);
/* Drop our extra ref: */
gst_buffer_unref (buf);
verify_test_buffer (out_buf);
cleanup_rglimiter (element);
}
GST_END_TEST;
Suite *
rglimiter_suite (void)
{
Suite *s = suite_create ("rglimiter");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_no_buffer);
tcase_add_test (tc_chain, test_disabled);
tcase_add_test (tc_chain, test_limiting);
return s;
}
int
main (int argc, char **argv)
{
gint nf;
Suite *s = rglimiter_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_ENV);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}

573
tests/check/elements/rgvolume.c Normal file
View file

@ -0,0 +1,573 @@
/* GStreamer ReplayGain volume adjustment
*
* Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
*
* rgvolume.c: Unit test for the rgvolume element
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#include <gst/check/gstcheck.h>
#include <math.h>
GList *buffers = NULL;
GList *events = NULL;
/* For ease of programming we use globals to keep refs for our floating src and
* sink pads we create; otherwise we always have to do get_pad, get_peer, and
* then remove references in every test function */
static GstPad *mysrcpad, *mysinkpad;
#define RG_VOLUME_CAPS_TEMPLATE_STRING \
"audio/x-raw-float, " \
"width = (int) 32, " \
"endianness = (int) BYTE_ORDER, " \
"channels = (int) [ 1, MAX ], " \
"rate = (int) [ 1, MAX ]"
static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (RG_VOLUME_CAPS_TEMPLATE_STRING)
);
static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (RG_VOLUME_CAPS_TEMPLATE_STRING)
);
/* gstcheck sets up a chain function that appends buffers to a global list.
* This is our equivalent of that for event handling. */
static gboolean
event_func (GstPad * pad, GstEvent * event)
{
events = g_list_append (events, event);
return TRUE;
}
GstElement *
setup_rgvolume ()
{
GstElement *element;
GST_DEBUG ("setup_rgvolume");
element = gst_check_setup_element ("rgvolume");
mysrcpad = gst_check_setup_src_pad (element, &srctemplate, NULL);
mysinkpad = gst_check_setup_sink_pad (element, &sinktemplate, NULL);
/* Capture events, to test tag filtering behavior: */
gst_pad_set_event_function (mysinkpad, event_func);
gst_pad_set_active (mysrcpad, TRUE);
gst_pad_set_active (mysinkpad, TRUE);
return element;
}
void
cleanup_rgvolume (GstElement * element)
{
GST_DEBUG ("cleanup_rgvolume");
g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
g_list_free (buffers);
buffers = NULL;
g_list_foreach (events, (GFunc) gst_mini_object_unref, NULL);
g_list_free (events);
events = NULL;
gst_pad_set_active (mysrcpad, FALSE);
gst_pad_set_active (mysinkpad, FALSE);
gst_check_teardown_src_pad (element);
gst_check_teardown_sink_pad (element);
gst_check_teardown_element (element);
}
static void
set_playing_state (GstElement * element)
{
fail_unless (gst_element_set_state (element,
GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
"Could not set state to PLAYING");
}
static void
set_null_state (GstElement * element)
{
fail_unless (gst_element_set_state (element,
GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS,
"Could not set state to NULL");
}
static void
send_eos_event (GstElement * element)
{
GstEvent *event = gst_event_new_eos ();
fail_unless (g_list_length (events) == 0);
fail_unless (gst_pad_push_event (mysrcpad, event),
"Pushing EOS event failed");
fail_unless (g_list_length (events) == 1);
fail_unless (events->data == event);
gst_mini_object_unref ((GstMiniObject *) events->data);
events = g_list_remove (events, event);
}
static GstEvent *
send_tag_event (GstElement * element, GstEvent * event)
{
g_return_val_if_fail (event->type == GST_EVENT_TAG, NULL);
fail_unless (g_list_length (events) == 0);
fail_unless (gst_pad_push_event (mysrcpad, event),
"Pushing tag event failed");
if (g_list_length (events) == 0) {
/* Event got filtered out. */
event = NULL;
} else {
GstTagList *tag_list;
gdouble dummy;
event = events->data;
events = g_list_remove (events, event);
fail_unless (event->type == GST_EVENT_TAG);
gst_event_parse_tag (event, &tag_list);
/* The element is supposed to filter out ReplayGain related tags. */
fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN, &dummy),
"tag event still contains track gain tag");
fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK, &dummy),
"tag event still contains track peak tag");
fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN, &dummy),
"tag event still contains album gain tag");
fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK, &dummy),
"tag event still contains album peak tag");
}
return event;
}
static GstBuffer *
test_buffer_new (gfloat value)
{
GstBuffer *buf;
GstCaps *caps;
gfloat *data;
gint i;
buf = gst_buffer_new_and_alloc (8 * sizeof (gfloat));
data = (gfloat *) GST_BUFFER_DATA (buf);
for (i = 0; i < 8; i++)
data[i] = value;
caps = gst_caps_from_string ("audio/x-raw-float, "
"rate = 8000, channels = 1, endianess = BYTE_ORDER, width = 32");
gst_buffer_set_caps (buf, caps);
gst_caps_unref (caps);
ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
return buf;
}
#define MATCH_GAIN(g1, g2) ((g1 < g2 + 1e-6) && (g2 < g1 + 1e-6))
static void
fail_unless_target_gain (GstElement * element, gdouble expected_gain)
{
gdouble prop_gain;
g_object_get (element, "target-gain", &prop_gain, NULL);
fail_unless (MATCH_GAIN (prop_gain, expected_gain),
"Target gain is %.2f dB, expected %.2f dB", prop_gain, expected_gain);
}
static void
fail_unless_result_gain (GstElement * element, gdouble expected_gain)
{
GstBuffer *input_buf, *output_buf;
gfloat input_sample, output_sample;
gdouble gain, prop_gain;
gboolean is_passthrough, expect_passthrough;
gint i;
fail_unless (g_list_length (buffers) == 0);
input_sample = 1.0;
input_buf = test_buffer_new (input_sample);
/* We keep an extra reference to detect passthrough mode. */
gst_buffer_ref (input_buf);
/* Pushing steals a reference. */
fail_unless (gst_pad_push (mysrcpad, input_buf) == GST_FLOW_OK);
gst_buffer_unref (input_buf);
/* The output buffer ends up on the global buffer list. */
fail_unless (g_list_length (buffers) == 1);
output_buf = buffers->data;
fail_if (output_buf == NULL);
buffers = g_list_remove (buffers, output_buf);
ASSERT_BUFFER_REFCOUNT (output_buf, "output_buf", 1);
fail_unless_equals_int (GST_BUFFER_SIZE (output_buf), 8 * sizeof (gfloat));
output_sample = *((gfloat *) GST_BUFFER_DATA (output_buf));
fail_if (output_sample == 0.0, "First output sample is zero");
for (i = 1; i < 8; i++) {
gfloat output = ((gfloat *) GST_BUFFER_DATA (output_buf))[i];
fail_unless (output_sample == output, "Output samples not uniform");
};
gain = 20. * log10 (output_sample / input_sample);
fail_unless (MATCH_GAIN (gain, expected_gain),
"Applied gain is %.2f dB, expected %.2f dB", gain, expected_gain);
g_object_get (element, "result-gain", &prop_gain, NULL);
fail_unless (MATCH_GAIN (prop_gain, expected_gain),
"Result gain is %.2f dB, expected %.2f dB", prop_gain, expected_gain);
is_passthrough = (output_buf == input_buf);
expect_passthrough = MATCH_GAIN (expected_gain, +0.00);
fail_unless (is_passthrough == expect_passthrough,
expect_passthrough
? "Expected operation in passthrough mode"
: "Incorrect passthrough behaviour");
gst_buffer_unref (output_buf);
}
static void
fail_unless_gain (GstElement * element, gdouble expected_gain)
{
fail_unless_target_gain (element, expected_gain);
fail_unless_result_gain (element, expected_gain);
}
/* Start of tests. */
GST_START_TEST (test_no_buffer)
{
GstElement *element = setup_rgvolume ();
set_playing_state (element);
set_null_state (element);
set_playing_state (element);
send_eos_event (element);
cleanup_rgvolume (element);
}
GST_END_TEST;
GST_START_TEST (test_events)
{
GstElement *element = setup_rgvolume ();
GstEvent *event;
GstEvent *new_event;
GstTagList *tag_list;
gchar *artist;
set_playing_state (element);
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_GAIN, +4.95, GST_TAG_TRACK_PEAK, 0.59463,
GST_TAG_ALBUM_GAIN, -1.54, GST_TAG_ALBUM_PEAK, 0.693415,
GST_TAG_ARTIST, "Foobar", NULL);
event = gst_event_new_tag (tag_list);
new_event = send_tag_event (element, event);
/* Expect the element to modify the writable event. */
fail_unless (event == new_event, "Writable tag event not reused");
gst_event_parse_tag (new_event, &tag_list);
fail_unless (gst_tag_list_get_string (tag_list, GST_TAG_ARTIST, &artist));
fail_unless (g_str_equal (artist, "Foobar"));
g_free (artist);
gst_event_unref (new_event);
/* Same as above, but with a non-writable event. */
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_GAIN, +4.95, GST_TAG_TRACK_PEAK, 0.59463,
GST_TAG_ALBUM_GAIN, -1.54, GST_TAG_ALBUM_PEAK, 0.693415,
GST_TAG_ARTIST, "Foobar", NULL);
event = gst_event_new_tag (tag_list);
/* Holding an extra ref makes the event unwritable: */
gst_event_ref (event);
new_event = send_tag_event (element, event);
fail_unless (event != new_event, "Unwritable tag event reused");
gst_event_parse_tag (new_event, &tag_list);
fail_unless (gst_tag_list_get_string (tag_list, GST_TAG_ARTIST, &artist));
fail_unless (g_str_equal (artist, "Foobar"));
g_free (artist);
gst_event_unref (event);
gst_event_unref (new_event);
cleanup_rgvolume (element);
}
GST_END_TEST;
GST_START_TEST (test_simple)
{
GstElement *element = setup_rgvolume ();
GstTagList *tag_list;
g_object_set (element, "album-mode", FALSE, "headroom", +0.00,
"pre-amp", -6.00, "fallback-gain", +1.23, NULL);
set_playing_state (element);
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_GAIN, -3.45, GST_TAG_TRACK_PEAK, 1.0,
GST_TAG_ALBUM_GAIN, +2.09, GST_TAG_ALBUM_PEAK, 1.0, NULL);
fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
fail_unless_gain (element, -9.45); /* pre-amp + track gain */
send_eos_event (element);
g_object_set (element, "album-mode", TRUE, NULL);
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_GAIN, -3.45, GST_TAG_TRACK_PEAK, 1.0,
GST_TAG_ALBUM_GAIN, +2.09, GST_TAG_ALBUM_PEAK, 1.0, NULL);
fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
fail_unless_gain (element, -3.91); /* pre-amp + album gain */
/* Switching back to track mode in the middle of a stream: */
g_object_set (element, "album-mode", FALSE, NULL);
fail_unless_gain (element, -9.45); /* pre-amp + track gain */
send_eos_event (element);
cleanup_rgvolume (element);
}
GST_END_TEST;
/* If there are no gain tags at all, the fallback gain is used. */
GST_START_TEST (test_fallback_gain)
{
GstElement *element = setup_rgvolume ();
GstTagList *tag_list;
/* First some track where fallback does _not_ apply. */
g_object_set (element, "album-mode", FALSE, "headroom", 10.00,
"pre-amp", -6.00, "fallback-gain", -3.00, NULL);
set_playing_state (element);
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_GAIN, +3.5, GST_TAG_TRACK_PEAK, 1.0,
GST_TAG_ALBUM_GAIN, -0.5, GST_TAG_ALBUM_PEAK, 1.0, NULL);
fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
fail_unless_gain (element, -2.50); /* pre-amp + track gain */
send_eos_event (element);
/* Now a track completely missing tags. */
fail_unless_gain (element, -9.00); /* pre-amp + fallback-gain */
/* Changing the fallback gain in the middle of a stream, going to pass-through
* mode: */
g_object_set (element, "fallback-gain", +6.00, NULL);
fail_unless_gain (element, +0.00); /* pre-amp + fallback-gain */
send_eos_event (element);
/* Verify that result gain is set to +0.00 with pre-amp + fallback-gain >
* +0.00 and no headroom. */
g_object_set (element, "fallback-gain", +12.00, "headroom", +0.00, NULL);
fail_unless_target_gain (element, +6.00); /* pre-amp + fallback-gain */
fail_unless_result_gain (element, +0.00);
send_eos_event (element);
cleanup_rgvolume (element);
}
GST_END_TEST;
/* If album gain is to be preferred but not available, the track gain is to be
* taken instead. */
GST_START_TEST (test_fallback_track)
{
GstElement *element = setup_rgvolume ();
GstTagList *tag_list;
g_object_set (element, "album-mode", TRUE, "headroom", +0.00,
"pre-amp", -6.00, "fallback-gain", +1.23, NULL);
set_playing_state (element);
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_GAIN, +2.11, GST_TAG_TRACK_PEAK, 1.0, NULL);
fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
fail_unless_gain (element, -3.89); /* pre-amp + track gain */
send_eos_event (element);
cleanup_rgvolume (element);
}
GST_END_TEST;
/* If track gain is to be preferred but not available, the album gain is to be
* taken instead. */
GST_START_TEST (test_fallback_album)
{
GstElement *element = setup_rgvolume ();
GstTagList *tag_list;
g_object_set (element, "album-mode", FALSE, "headroom", +0.00,
"pre-amp", -6.00, "fallback-gain", +1.23, NULL);
set_playing_state (element);
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_ALBUM_GAIN, +3.73, GST_TAG_ALBUM_PEAK, 1.0, NULL);
fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
fail_unless_gain (element, -2.27); /* pre-amp + album gain */
send_eos_event (element);
cleanup_rgvolume (element);
}
GST_END_TEST;
GST_START_TEST (test_headroom)
{
GstElement *element = setup_rgvolume ();
GstTagList *tag_list;
g_object_set (element, "album-mode", FALSE, "headroom", +0.00,
"pre-amp", +0.00, "fallback-gain", +1.23, NULL);
set_playing_state (element);
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_GAIN, +3.50, GST_TAG_TRACK_PEAK, 1.0, NULL);
fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
fail_unless_target_gain (element, +3.50); /* pre-amp + track gain */
fail_unless_result_gain (element, +0.00);
send_eos_event (element);
g_object_set (element, "headroom", +2.00, NULL);
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_GAIN, +9.18, GST_TAG_TRACK_PEAK, 0.687149, NULL);
fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
fail_unless_target_gain (element, +9.18); /* pre-amp + track gain */
/* Result is 20. * log10 (1. / peak) + headroom. */
fail_unless_result_gain (element, 5.2589816238303335);
send_eos_event (element);
g_object_set (element, "album-mode", TRUE, NULL);
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_ALBUM_GAIN, +5.50, GST_TAG_ALBUM_PEAK, 1.0, NULL);
fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
fail_unless_target_gain (element, +5.50); /* pre-amp + album gain */
fail_unless_result_gain (element, +2.00); /* headroom */
send_eos_event (element);
cleanup_rgvolume (element);
}
GST_END_TEST;
GST_START_TEST (test_reference_level)
{
GstElement *element = setup_rgvolume ();
GstTagList *tag_list;
g_object_set (element,
"album-mode", FALSE,
"headroom", +0.00, "pre-amp", +0.00, "fallback-gain", +1.23, NULL);
set_playing_state (element);
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_GAIN, 0.00, GST_TAG_TRACK_PEAK, 0.2,
GST_TAG_REFERENCE_LEVEL, 83., NULL);
fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
/* Because our authorative reference is 89 dB, we bump it up by +6 dB. */
fail_unless_gain (element, +6.00); /* pre-amp + track gain */
send_eos_event (element);
g_object_set (element, "album-mode", TRUE, NULL);
/* Same as above, but with album gain. */
tag_list = gst_tag_list_new ();
gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
GST_TAG_TRACK_GAIN, 1.23, GST_TAG_TRACK_PEAK, 0.1,
GST_TAG_ALBUM_GAIN, 0.00, GST_TAG_ALBUM_PEAK, 0.2,
GST_TAG_REFERENCE_LEVEL, 83., NULL);
fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
fail_unless_gain (element, +6.00); /* pre-amp + album gain */
cleanup_rgvolume (element);
}
GST_END_TEST;
Suite *
rgvolume_suite (void)
{
Suite *s = suite_create ("rgvolume");
TCase *tc_chain = tcase_create ("general");
suite_add_tcase (s, tc_chain);
tcase_add_test (tc_chain, test_no_buffer);
tcase_add_test (tc_chain, test_events);
tcase_add_test (tc_chain, test_simple);
tcase_add_test (tc_chain, test_fallback_gain);
tcase_add_test (tc_chain, test_fallback_track);
tcase_add_test (tc_chain, test_fallback_album);
tcase_add_test (tc_chain, test_headroom);
tcase_add_test (tc_chain, test_reference_level);
return s;
}
int
main (int argc, char **argv)
{
gint nf;
Suite *s = rgvolume_suite ();
SRunner *sr = srunner_create (s);
gst_check_init (&argc, &argv);
srunner_run_all (sr, CK_ENV);
nf = srunner_ntests_failed (sr);
srunner_free (sr);
return nf;
}