Add ReplayGain analysis element (#357069).

Original commit message from CVS:
Patch by: René Stadler  <mail at renestadler de>
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/replaygain/Makefile.am:
* gst/replaygain/gstrganalysis.c: (gst_rg_analysis_base_init),
(gst_rg_analysis_class_init), (gst_rg_analysis_init),
(gst_rg_analysis_set_property), (gst_rg_analysis_get_property),
(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), (plugin_init):
* gst/replaygain/gstrganalysis.h:
* gst/replaygain/rganalysis.c: (yule_filter), (butter_filter),
(apply_filters), (reset_filters), (accumulator_add),
(accumulator_clear), (accumulator_result), (rg_analysis_new),
(rg_analysis_set_sample_rate), (rg_analysis_destroy),
(rg_analysis_analyze_mono_float),
(rg_analysis_analyze_stereo_float),
(rg_analysis_analyze_mono_int16),
(rg_analysis_analyze_stereo_int16), (rg_analysis_analyze),
(rg_analysis_track_result), (rg_analysis_album_result),
(rg_analysis_reset_album), (rg_analysis_reset):
* gst/replaygain/rganalysis.h:
Add ReplayGain analysis element (#357069).
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/rganalysis.c: (get_expected_gain),
(setup_rganalysis), (cleanup_rganalysis), (set_playing_state),
(send_eos_event), (send_tag_event), (poll_eos), (poll_tags),
(fail_unless_track_gain), (fail_unless_track_peak),
(fail_unless_album_gain), (fail_unless_album_peak),
(fail_if_track_tags), (fail_if_album_tags),
(fail_unless_num_tracks), (test_buffer_const_float_mono),
(test_buffer_const_float_stereo), (test_buffer_const_int16_mono),
(test_buffer_const_int16_stereo), (test_buffer_square_float_mono),
(test_buffer_square_float_stereo), (test_buffer_square_int16_mono),
(test_buffer_square_int16_stereo), (push_buffer), (GST_START_TEST),
(rganalysis_suite), (main):
Unit tests for the new replaygain element.
This commit is contained in:
René Stadler 2006-10-06 15:56:01 +00:00 committed by Tim-Philipp Müller
parent 09624a5894
commit 30c429a214
13 changed files with 3545 additions and 0 deletions

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@ -1,3 +1,50 @@
2006-10-06 Tim-Philipp Müller <tim at centricular dot net>
Patch by: René Stadler <mail at renestadler de>
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* gst/replaygain/Makefile.am:
* gst/replaygain/gstrganalysis.c: (gst_rg_analysis_base_init),
(gst_rg_analysis_class_init), (gst_rg_analysis_init),
(gst_rg_analysis_set_property), (gst_rg_analysis_get_property),
(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), (plugin_init):
* gst/replaygain/gstrganalysis.h:
* gst/replaygain/rganalysis.c: (yule_filter), (butter_filter),
(apply_filters), (reset_filters), (accumulator_add),
(accumulator_clear), (accumulator_result), (rg_analysis_new),
(rg_analysis_set_sample_rate), (rg_analysis_destroy),
(rg_analysis_analyze_mono_float),
(rg_analysis_analyze_stereo_float),
(rg_analysis_analyze_mono_int16),
(rg_analysis_analyze_stereo_int16), (rg_analysis_analyze),
(rg_analysis_track_result), (rg_analysis_album_result),
(rg_analysis_reset_album), (rg_analysis_reset):
* gst/replaygain/rganalysis.h:
Add ReplayGain analysis element (#357069).
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/rganalysis.c: (get_expected_gain),
(setup_rganalysis), (cleanup_rganalysis), (set_playing_state),
(send_eos_event), (send_tag_event), (poll_eos), (poll_tags),
(fail_unless_track_gain), (fail_unless_track_peak),
(fail_unless_album_gain), (fail_unless_album_peak),
(fail_if_track_tags), (fail_if_album_tags),
(fail_unless_num_tracks), (test_buffer_const_float_mono),
(test_buffer_const_float_stereo), (test_buffer_const_int16_mono),
(test_buffer_const_int16_stereo), (test_buffer_square_float_mono),
(test_buffer_square_float_stereo), (test_buffer_square_int16_mono),
(test_buffer_square_int16_stereo), (push_buffer), (GST_START_TEST),
(rganalysis_suite), (main):
Unit tests for the new replaygain element.
2006-10-06 Wim Taymans <wim@fluendo.com> 2006-10-06 Wim Taymans <wim@fluendo.com>
* ext/faad/gstfaad.c: (gst_faad_setcaps), (gst_faad_chain), * ext/faad/gstfaad.c: (gst_faad_setcaps), (gst_faad_chain),

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@ -82,6 +82,7 @@ GST_PLUGINS_ALL="\
h264parse \ h264parse \
modplug \ modplug \
nsf \ nsf \
replaygain \
spectrum \ spectrum \
speed \ speed \
qtdemux \ qtdemux \
@ -787,6 +788,7 @@ gst/h264parse/Makefile
gst/modplug/Makefile gst/modplug/Makefile
gst/modplug/libmodplug/Makefile gst/modplug/libmodplug/Makefile
gst/nsf/Makefile gst/nsf/Makefile
gst/replaygain/Makefile
gst/spectrum/Makefile gst/spectrum/Makefile
gst/speed/Makefile gst/speed/Makefile
gst/qtdemux/Makefile gst/qtdemux/Makefile

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@ -91,6 +91,7 @@ EXTRA_HFILES = \
$(top_srcdir)/ext/wavpack/gstwavpackdec.h \ $(top_srcdir)/ext/wavpack/gstwavpackdec.h \
$(top_srcdir)/ext/wavpack/gstwavpackenc.h \ $(top_srcdir)/ext/wavpack/gstwavpackenc.h \
$(top_srcdir)/ext/wavpack/gstwavpackparse.h \ $(top_srcdir)/ext/wavpack/gstwavpackparse.h \
$(top_srcdir)/gst/replaygain/gstrganalysis.h \
$(top_srcdir)/gst/videocrop/gstvideocrop.h $(top_srcdir)/gst/videocrop/gstvideocrop.h
# Images to copy into HTML directory. # Images to copy into HTML directory.

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@ -14,6 +14,7 @@
<title>gst-plugins-bad Elements</title> <title>gst-plugins-bad Elements</title>
<xi:include href="xml/element-dfbvideosink.xml" /> <xi:include href="xml/element-dfbvideosink.xml" />
<xi:include href="xml/element-rganalysis.xml" />
<xi:include href="xml/element-trm.xml" /> <xi:include href="xml/element-trm.xml" />
<xi:include href="xml/element-videocrop.xml" /> <xi:include href="xml/element-videocrop.xml" />
<xi:include href="xml/element-wavpackdec.xml" /> <xi:include href="xml/element-wavpackdec.xml" />
@ -37,6 +38,7 @@
<xi:include href="xml/plugin-musicbrainz.xml" /> <xi:include href="xml/plugin-musicbrainz.xml" />
<xi:include href="xml/plugin-neon.xml" /> <xi:include href="xml/plugin-neon.xml" />
<xi:include href="xml/plugin-qtdemux.xml" /> <xi:include href="xml/plugin-qtdemux.xml" />
<xi:include href="xml/plugin-replaygain.xml" />
<xi:include href="xml/plugin-sdlvideosink.xml" /> <xi:include href="xml/plugin-sdlvideosink.xml" />
<xi:include href="xml/plugin-sdl.xml" /> <xi:include href="xml/plugin-sdl.xml" />
<xi:include href="xml/plugin-spectrum.xml" /> <xi:include href="xml/plugin-spectrum.xml" />

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@ -6,6 +6,14 @@ GstDfbVideoSink
GstDfbVideoSinkClass GstDfbVideoSinkClass
</SECTION> </SECTION>
<SECTION>
<FILE>element-rganalysis</FILE>
GstRgAnalysis
<TITLE>rganalysis</TITLE>
<SUBSECTION Standard>
GstRgAnalysisClass
</SECTION>
<SECTION> <SECTION>
<FILE>element-trm</FILE> <FILE>element-trm</FILE>
GstTRM GstTRM

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@ -0,0 +1,13 @@
plugin_LTLIBRARIES = libgstreplaygain.la
libgstreplaygain_la_SOURCES = \
gstrganalysis.c \
rganalysis.c
libgstreplaygain_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS)
libgstreplaygain_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS)
libgstreplaygain_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
# headers we need but don't want installed
noinst_HEADERS = \
gstrganalysis.h \
rganalysis.h

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@ -0,0 +1,686 @@
/* GStreamer ReplayGain analysis
*
* Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
*
* gstrganalysis.c: Element that performs the ReplayGain analysis
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
/**
* SECTION:element-rganalysis
*
* <refsect2>
* <para>
* GstRgAnalysis analyzes raw audio sample data in accordance with the
* proposed <ulink url="http://replaygain.org">ReplayGain
* standard</ulink> for calculating the ideal replay gain for music
* tracks and albums. The element is designed as a pass-through
* filter that never modifies any data. As it receives an EOS event,
* it finalizes the ongoing analysis and generates a tag list
* containing the results. It is sent downstream with a TAG event and
* posted on the message bus with a TAG message. The EOS event is
* forwarded as normal afterwards. Result tag lists at least contain
* the tags #GST_TAG_TRACK_GAIN and #GST_TAG_TRACK_PEAK.
* </para>
* <title>Album processing</title>
* <para>
* Analyzing several streams sequentially and assigning them a common
* result gain is known as "album processing". If this gain is used
* during playback (by switching to "album mode"), all tracks receive
* the same amplification. This keeps the relative volume levels
* between the tracks intact. To enable this, set the <link
* linkend="GstRgAnalysis--num-tracks">num-tracks</link> property to
* the number of streams that will be processed as album tracks.
* Every time an EOS event is received, the value of this property
* will be decremented by one. As it reaches zero, it is assumed that
* the last track of the album finished. The tag list for the final
* stream will contain the additional tags #GST_TAG_ALBUM_GAIN and
* #GST_TAG_ALBUM_PEAK. All other streams just get the two track tags
* posted because the values for the album tags are not known before
* all tracks are analyzed. Applications need to make sure that the
* album gain and peak values are also associated with the other
* tracks when storing the results. It is thus a bit more complex to
* implement, but should not be avoided since the album gain is
* generally more valuable for use during playback than the track
* gain.
* </para>
* <title>Skipping processing</title>
* <para>
* For assisting transcoder/converter applications, the element can
* silently skip the processing of streams that already contain the
* necessary meta data tags. Data will flow as usual but the element
* will not consume CPU time and will not generate result tags. To
* enable possible skipping, set the <link
* linkend="GstRgAnalysis--forced">forced</link> property to #FALSE.
* If used in conjunction with album processing, the element will skip
* the number of remaining album tracks if a full set of tags is found
* for the first track. If a subsequent track of the album is missing
* tags, processing cannot start again. If this is undesired, your
* application has to scan all files beforehand and enable forcing of
* processing if needed.
* </para>
* <title>Tips</title>
* <itemizedlist>
* <listitem><para>
* Because the generated metadata tags become available at the end of
* streams, downstream muxer and encoder elements are normally unable
* to save them in their output since they generally save metadata in
* the file header. Therefore, it is often necessary that
* applications read the results in a bus event handler for the tag
* message. Obtaining the values this way is always needed for album
* processing since the album gain and peak values need to be
* associated with all tracks of an album, not just the last one.
* </para></listitem>
* <listitem><para>
* To perform album processing, the element has to preserve data
* between streams. This cannot survive a state change to the NULL or
* READY state. If you change your pipeline's state to NULL or READY
* between tracks, lock the rganalysis element's state using
* gst_element_set_locked_state() when it is in PAUSED or PLAYING. As
* with any other element, don't forget to unlock it again and set it
* to the NULL state before dropping the last reference.
* </para></listitem>
* <listitem><para>
* If the total number of album tracks is unknown beforehand, set the
* num-tracks property to some large value like #G_MAXINT (or set it
* to >= 2 before each track starts). Before the last track ends, set
* the property value to 1.
* </para></listitem>
* </itemizedlist>
* <title>Compliance</title>
* <para>
* Analyzing the ReplayGain pink noise reference waveform will compute
* a result of +6.00dB instead of the expected 0.00dB because the
* default reference level is 89dB. To obtain values as lined out in
* the original proposal of ReplayGain, set the <link
* linkend="GstRgAnalysis--reference-level">reference-level</link>
* property to 83. Almost all software uses 89dB as a reference
* however, which works against the tendency of the algorithm to
* advise to drastically lower the volume of music with a highly
* compressed dynamic range and high average output levels. This
* tendency is normally to be fought during playback (if wanted), by
* using a default pre-amp value of at least +6.00dB. At one point,
* the majority of analyzer implementations switched to 89dB which
* moved this adjustment to the analyzing/metadata writing process.
* This change has been acknowledged by the author of the ReplayGain
* proposal, however at the time of this writing, the webpage is still
* not updated.
* </para>
* <title>Example launch lines</title>
* <para>Analyze a simple test waveform:</para>
* <programlisting>
* gst-launch -t audiotestsrc wave=sine num-buffers=512 ! rganalysis ! fakesink
* </programlisting>
* <para>Analyze a given file:</para>
* <programlisting>
* gst-launch -t filesrc location="Some file.ogg" ! decodebin ! audioconvert ! audioresample ! rganalysis ! fakesink
* </programlisting>
* <para>Analyze the pink noise reference file:</para>
* <programlisting>
* gst-launch -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav ! wavparse ! rganalysis ! fakesink
* </programlisting>
* <title>Acknowledgements</title>
* <para>
* This element is based on code used in the <ulink
* url="http://sjeng.org/vorbisgain.html">vorbisgain</ulink> program
* and many others. The relevant parts are copyrighted by David
* Robinson, Glen Sawyer and Frank Klemm.
* </para>
* </refsect2>
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include <string.h>
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include "gstrganalysis.h"
GST_DEBUG_CATEGORY_STATIC (gst_rg_analysis_debug);
#define GST_CAT_DEFAULT gst_rg_analysis_debug
static const GstElementDetails rganalysis_details = {
"ReplayGain analysis",
"Filter/Analyzer/Audio",
"Perform the ReplayGain analysis",
"Ren\xc3\xa9 Stadler <mail@renestadler.de>"
};
/* Default property value. */
#define FORCED_DEFAULT TRUE
enum
{
PROP_0,
PROP_NUM_TRACKS,
PROP_FORCED,
PROP_REFERENCE_LEVEL
};
/* The ReplayGain algorithm is intended for use with mono and stereo
* audio. The used implementation has filter coefficients for the
* "usual" sample rates in the 8000 to 48000 Hz range. */
#define REPLAY_GAIN_CAPS \
"channels = (int) { 1, 2 }, " \
"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, " \
"44100, 48000 }"
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
"width = (int) 32, " "endianness = (int) BYTE_ORDER, "
REPLAY_GAIN_CAPS "; "
"audio/x-raw-int, "
"width = (int) 16, " "depth = (int) [ 1, 16 ], "
"signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
REPLAY_GAIN_CAPS));
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
"width = (int) 32, " "endianness = (int) BYTE_ORDER, "
REPLAY_GAIN_CAPS "; "
"audio/x-raw-int, "
"width = (int) 16, " "depth = (int) [ 1, 16 ], "
"signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
REPLAY_GAIN_CAPS));
GST_BOILERPLATE (GstRgAnalysis, gst_rg_analysis, GstBaseTransform,
GST_TYPE_BASE_TRANSFORM);
static void gst_rg_analysis_class_init (GstRgAnalysisClass * klass);
static void gst_rg_analysis_init (GstRgAnalysis * filter,
GstRgAnalysisClass * gclass);
static void gst_rg_analysis_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_rg_analysis_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_rg_analysis_start (GstBaseTransform * base);
static gboolean gst_rg_analysis_set_caps (GstBaseTransform * base,
GstCaps * incaps, GstCaps * outcaps);
static GstFlowReturn gst_rg_analysis_transform_ip (GstBaseTransform * base,
GstBuffer * buf);
static gboolean gst_rg_analysis_event (GstBaseTransform * base,
GstEvent * event);
static gboolean gst_rg_analysis_stop (GstBaseTransform * base);
static void gst_rg_analysis_handle_tags (GstRgAnalysis * filter,
const GstTagList * tag_list);
static void gst_rg_analysis_handle_eos (GstRgAnalysis * filter);
static gboolean gst_rg_analysis_track_result (GstRgAnalysis * filter,
GstTagList ** tag_list);
static gboolean gst_rg_analysis_album_result (GstRgAnalysis * filter,
GstTagList ** tag_list);
static void
gst_rg_analysis_base_init (gpointer g_class)
{
GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&src_factory));
gst_element_class_add_pad_template (element_class,
gst_static_pad_template_get (&sink_factory));
gst_element_class_set_details (element_class, &rganalysis_details);
GST_DEBUG_CATEGORY_INIT (gst_rg_analysis_debug, "rganalysis", 0,
"ReplayGain analysis element");
}
static void
gst_rg_analysis_class_init (GstRgAnalysisClass * klass)
{
GObjectClass *gobject_class;
GstBaseTransformClass *trans_class;
gobject_class = (GObjectClass *) klass;
gobject_class->set_property = gst_rg_analysis_set_property;
gobject_class->get_property = gst_rg_analysis_get_property;
g_object_class_install_property (gobject_class, PROP_NUM_TRACKS,
g_param_spec_int ("num-tracks", "Number of album tracks",
"Number of remaining tracks in the album",
0, G_MAXINT, 0, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_FORCED,
g_param_spec_boolean ("forced", "Force processing",
"Analyze streams even when ReplayGain tags exist",
FORCED_DEFAULT, G_PARAM_READWRITE));
g_object_class_install_property (gobject_class, PROP_REFERENCE_LEVEL,
g_param_spec_double ("reference-level", "Reference level",
"Reference level in dB (83.0 for original proposal)",
0.0, G_MAXDOUBLE, RG_REFERENCE_LEVEL, G_PARAM_READWRITE));
trans_class = (GstBaseTransformClass *) klass;
trans_class->start = GST_DEBUG_FUNCPTR (gst_rg_analysis_start);
trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_rg_analysis_set_caps);
trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_analysis_transform_ip);
trans_class->event = GST_DEBUG_FUNCPTR (gst_rg_analysis_event);
trans_class->stop = GST_DEBUG_FUNCPTR (gst_rg_analysis_stop);
trans_class->passthrough_on_same_caps = TRUE;
}
static void
gst_rg_analysis_init (GstRgAnalysis * filter, GstRgAnalysisClass * gclass)
{
filter->num_tracks = 0;
filter->forced = FORCED_DEFAULT;
filter->reference_level = RG_REFERENCE_LEVEL;
filter->ctx = NULL;
filter->analyze = NULL;
}
static void
gst_rg_analysis_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstRgAnalysis *filter = GST_RG_ANALYSIS (object);
switch (prop_id) {
case PROP_NUM_TRACKS:
filter->num_tracks = g_value_get_int (value);
break;
case PROP_FORCED:
filter->forced = g_value_get_boolean (value);
break;
case PROP_REFERENCE_LEVEL:
filter->reference_level = g_value_get_double (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_rg_analysis_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec)
{
GstRgAnalysis *filter = GST_RG_ANALYSIS (object);
switch (prop_id) {
case PROP_NUM_TRACKS:
g_value_set_int (value, filter->num_tracks);
break;
case PROP_FORCED:
g_value_set_boolean (value, filter->forced);
break;
case PROP_REFERENCE_LEVEL:
g_value_set_double (value, filter->reference_level);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static gboolean
gst_rg_analysis_start (GstBaseTransform * base)
{
GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
filter->ignore_tags = FALSE;
filter->skip = FALSE;
filter->has_track_gain = FALSE;
filter->has_track_peak = FALSE;
filter->has_album_gain = FALSE;
filter->has_album_peak = FALSE;
filter->ctx = rg_analysis_new ();
filter->analyze = NULL;
GST_DEBUG_OBJECT (filter, "Started");
return TRUE;
}
static gboolean
gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
GstCaps * out_caps)
{
GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
GstStructure *structure;
const gchar *mime_type;
gint n_channels, sample_rate, sample_bit_size, sample_size;
g_return_val_if_fail (filter->ctx != NULL, FALSE);
GST_DEBUG_OBJECT (filter,
"set_caps in %" GST_PTR_FORMAT " out %" GST_PTR_FORMAT,
in_caps, out_caps);
structure = gst_caps_get_structure (in_caps, 0);
mime_type = gst_structure_get_name (structure);
if (!gst_structure_get_int (structure, "width", &sample_bit_size)
|| !gst_structure_get_int (structure, "channels", &n_channels)
|| !gst_structure_get_int (structure, "rate", &sample_rate))
goto invalid_format;
if (!rg_analysis_set_sample_rate (filter->ctx, sample_rate))
goto invalid_format;
if (sample_bit_size % 8 != 0)
goto invalid_format;
sample_size = sample_bit_size / 8;
if (strcmp (mime_type, "audio/x-raw-float") == 0) {
if (sample_size != sizeof (gfloat))
goto invalid_format;
/* The depth is not variable for float formats of course. It just
* makes the transform function nice and simple if the
* rg_analysis_analyze_* functions have a common signature. */
filter->depth = sizeof (gfloat) * 8;
if (n_channels == 1)
filter->analyze = rg_analysis_analyze_mono_float;
else if (n_channels == 2)
filter->analyze = rg_analysis_analyze_stereo_float;
else
goto invalid_format;
} else if (strcmp (mime_type, "audio/x-raw-int") == 0) {
if (sample_size != sizeof (gint16))
goto invalid_format;
if (!gst_structure_get_int (structure, "depth", &filter->depth))
goto invalid_format;
if (filter->depth < 1 || filter->depth > 16)
goto invalid_format;
if (n_channels == 1)
filter->analyze = rg_analysis_analyze_mono_int16;
else if (n_channels == 2)
filter->analyze = rg_analysis_analyze_stereo_int16;
else
goto invalid_format;
} else {
goto invalid_format;
}
return TRUE;
/* Errors. */
invalid_format:
{
filter->analyze = NULL;
GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
("Invalid incoming caps: %" GST_PTR_FORMAT, in_caps), (NULL));
return FALSE;
}
}
static GstFlowReturn
gst_rg_analysis_transform_ip (GstBaseTransform * base, GstBuffer * buf)
{
GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
g_return_val_if_fail (filter->ctx != NULL, GST_FLOW_ERROR);
g_return_val_if_fail (filter->analyze != NULL, GST_FLOW_ERROR);
if (filter->skip)
return GST_FLOW_OK;
GST_DEBUG_OBJECT (filter, "Processing buffer of size %u",
GST_BUFFER_SIZE (buf));
filter->analyze (filter->ctx, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
filter->depth);
return GST_FLOW_OK;
}
static gboolean
gst_rg_analysis_event (GstBaseTransform * base, GstEvent * event)
{
GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
g_return_val_if_fail (filter->ctx != NULL, TRUE);
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_EOS:
{
GST_DEBUG_OBJECT (filter, "Received EOS event");
gst_rg_analysis_handle_eos (filter);
GST_DEBUG_OBJECT (filter, "Passing on EOS event");
break;
}
case GST_EVENT_TAG:
{
GstTagList *tag_list;
/* The reference to the tag list is borrowed. */
gst_event_parse_tag (event, &tag_list);
gst_rg_analysis_handle_tags (filter, tag_list);
break;
}
default:
break;
}
return TRUE;
}
static gboolean
gst_rg_analysis_stop (GstBaseTransform * base)
{
GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
g_return_val_if_fail (filter->ctx != NULL, FALSE);
rg_analysis_destroy (filter->ctx);
filter->ctx = NULL;
GST_DEBUG_OBJECT (filter, "Stopped");
return TRUE;
}
static void
gst_rg_analysis_handle_tags (GstRgAnalysis * filter,
const GstTagList * tag_list)
{
gboolean album_processing = (filter->num_tracks > 0);
gdouble dummy;
if (!album_processing)
filter->ignore_tags = FALSE;
if (filter->skip && album_processing) {
GST_INFO_OBJECT (filter, "Ignoring TAG event: Skipping album");
return;
} else if (filter->skip) {
GST_INFO_OBJECT (filter, "Ignoring TAG event: Skipping track");
return;
} else if (filter->ignore_tags) {
GST_INFO_OBJECT (filter, "Ignoring TAG event: Cannot skip anyways");
return;
}
filter->has_track_gain |= gst_tag_list_get_double (tag_list,
GST_TAG_TRACK_GAIN, &dummy);
filter->has_track_peak |= gst_tag_list_get_double (tag_list,
GST_TAG_TRACK_PEAK, &dummy);
filter->has_album_gain |= gst_tag_list_get_double (tag_list,
GST_TAG_ALBUM_GAIN, &dummy);
filter->has_album_peak |= gst_tag_list_get_double (tag_list,
GST_TAG_ALBUM_PEAK, &dummy);
if (!(filter->has_track_gain && filter->has_track_peak)) {
GST_INFO_OBJECT (filter, "Track tags not complete yet");
return;
}
if (album_processing && !(filter->has_album_gain && filter->has_album_peak)) {
GST_INFO_OBJECT (filter, "Album tags not complete yet");
return;
}
if (filter->forced) {
GST_INFO_OBJECT (filter,
"Existing tags are sufficient, but processing anyway (forced)");
return;
}
filter->skip = TRUE;
rg_analysis_reset (filter->ctx);
if (!album_processing)
GST_INFO_OBJECT (filter,
"Existing tags are sufficient, will not process this track");
else
GST_INFO_OBJECT (filter,
"Existing tags are sufficient, will not process this album");
}
static void
gst_rg_analysis_handle_eos (GstRgAnalysis * filter)
{
gboolean album_processing = (filter->num_tracks > 0);
gboolean album_finished = (filter->num_tracks == 1);
gboolean album_skipping = album_processing && filter->skip;
filter->has_track_gain = FALSE;
filter->has_track_peak = FALSE;
if (album_finished) {
filter->ignore_tags = FALSE;
filter->skip = FALSE;
filter->has_album_gain = FALSE;
filter->has_album_peak = FALSE;
} else if (!album_skipping) {
filter->skip = FALSE;
}
/* We might have just fully processed a track because it has
* incomplete tags. If we do album processing and allow skipping
* (not forced), prevent switching to skipping if a later track with
* full tags comes along: */
if (!filter->forced && album_processing && !album_finished)
filter->ignore_tags = TRUE;
if (!filter->skip) {
GstTagList *tag_list = NULL;
gboolean track_success;
gboolean album_success = FALSE;
track_success = gst_rg_analysis_track_result (filter, &tag_list);
if (album_finished)
album_success = gst_rg_analysis_album_result (filter, &tag_list);
else if (!album_processing)
rg_analysis_reset_album (filter->ctx);
if (track_success || album_success) {
GST_DEBUG_OBJECT (filter, "Posting tag list with results");
/* This steals our reference to the list: */
gst_element_found_tags_for_pad (GST_ELEMENT (filter),
GST_BASE_TRANSFORM_SRC_PAD (GST_BASE_TRANSFORM (filter)), tag_list);
}
}
if (album_processing) {
filter->num_tracks--;
if (!album_finished)
GST_INFO_OBJECT (filter, "Album not finished yet (num-tracks is now %u)",
filter->num_tracks);
else
GST_INFO_OBJECT (filter, "Album finished (num-tracks is now 0)");
}
if (album_processing)
g_object_notify (G_OBJECT (filter), "num-tracks");
}
static gboolean
gst_rg_analysis_track_result (GstRgAnalysis * filter, GstTagList ** tag_list)
{
gboolean track_success;
gdouble track_gain, track_peak;
track_success = rg_analysis_track_result (filter->ctx, &track_gain,
&track_peak);
if (track_success) {
track_gain += filter->reference_level - RG_REFERENCE_LEVEL;
GST_INFO_OBJECT (filter, "Track gain is %+.2f dB, peak %.6f", track_gain,
track_peak);
} else {
GST_INFO_OBJECT (filter, "Track was too short to analyze");
}
if (track_success) {
if (*tag_list == NULL)
*tag_list = gst_tag_list_new ();
gst_tag_list_add (*tag_list, GST_TAG_MERGE_APPEND,
GST_TAG_TRACK_PEAK, track_peak, GST_TAG_TRACK_GAIN, track_gain, NULL);
}
return track_success;
}
static gboolean
gst_rg_analysis_album_result (GstRgAnalysis * filter, GstTagList ** tag_list)
{
gboolean album_success;
gdouble album_gain, album_peak;
album_success = rg_analysis_album_result (filter->ctx, &album_gain,
&album_peak);
if (album_success) {
album_gain += filter->reference_level - RG_REFERENCE_LEVEL;
GST_INFO_OBJECT (filter, "Album gain is %+.2f dB, peak %.6f", album_gain,
album_peak);
} else {
GST_INFO_OBJECT (filter, "Album was too short to analyze");
}
if (album_success) {
if (*tag_list == NULL)
*tag_list = gst_tag_list_new ();
gst_tag_list_add (*tag_list, GST_TAG_MERGE_APPEND,
GST_TAG_ALBUM_PEAK, album_peak, GST_TAG_ALBUM_GAIN, album_gain, NULL);
}
return album_success;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
return gst_element_register (plugin, "rganalysis", GST_RANK_NONE,
GST_TYPE_RG_ANALYSIS);
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "replaygain",
"ReplayGain analysis", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME,
GST_PACKAGE_ORIGIN);

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@ -0,0 +1,83 @@
/* GStreamer ReplayGain analysis
*
* Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
*
* gstrganalysis.h: Element that performs the ReplayGain analysis
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#ifndef __GST_RG_ANALYSIS_H__
#define __GST_RG_ANALYSIS_H__
#include <gst/gst.h>
#include <gst/base/gstbasetransform.h>
#include "rganalysis.h"
G_BEGIN_DECLS
#define GST_TYPE_RG_ANALYSIS \
(gst_rg_analysis_get_type())
#define GST_RG_ANALYSIS(obj) \
(G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_ANALYSIS,GstRgAnalysis))
#define GST_RG_ANALYSIS_CLASS(klass) \
(G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_ANALYSIS,GstRgAnalysisClass))
#define GST_IS_RG_ANALYSIS(obj) \
(G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_ANALYSIS))
#define GST_IS_RG_ANALYSIS_CLASS(klass) \
(G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_ANALYSIS))
typedef struct _GstRgAnalysis GstRgAnalysis;
typedef struct _GstRgAnalysisClass GstRgAnalysisClass;
/**
* GstRgAnalysis:
*
* Opaque data structure.
*/
struct _GstRgAnalysis
{
GstBaseTransform element;
/*< private >*/
RgAnalysisCtx *ctx;
void (*analyze) (RgAnalysisCtx * ctx, gconstpointer data, gsize size,
guint depth);
gint depth;
/* Property values. */
guint num_tracks;
gdouble reference_level;
gboolean forced;
/* State machinery for skipping. */
gboolean ignore_tags;
gboolean skip;
gboolean has_track_gain;
gboolean has_track_peak;
gboolean has_album_gain;
gboolean has_album_peak;
};
struct _GstRgAnalysisClass
{
GstBaseTransformClass parent_class;
};
G_END_DECLS
#endif /* __GST_RG_ANALYSIS_H__ */

772
gst/replaygain/rganalysis.c Normal file
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@ -0,0 +1,772 @@
/* GStreamer ReplayGain analysis
*
* Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
* Copyright (C) 2001 David Robinson <David@Robinson.org>
* Glen Sawyer <glensawyer@hotmail.com>
*
* rganalysis.c: Analyze raw audio data in accordance with ReplayGain
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
/* Based on code with Copyright (C) 2001 David Robinson
* <David@Robinson.org> and Glen Sawyer <glensawyer@hotmail.com>,
* which is distributed under the LGPL as part of the vorbisgain
* program. The original code also mentions Frank Klemm
* (http://www.uni-jena.de/~pfk/mpp/) for having contributed lots of
* good code. Specifically, this is based on the file
* "gain_analysis.c" from vorbisgain version 0.34.
*/
/* Room for future improvement: Mono data is currently in fact copied
* to two channels which get processed normally. This means that mono
* input data is processed twice.
*/
/* Helpful information for understanding this code: The two IIR
* filters depend on previous input _and_ previous output samples (up
* to the filter's order number of samples). This explains the whole
* lot of memcpy'ing done in rg_analysis_analyze and why the context
* holds so many buffers.
*/
#include <math.h>
#include <string.h>
#include <glib.h>
#include "rganalysis.h"
#define YULE_ORDER 10
#define BUTTER_ORDER 2
/* Percentile which is louder than the proposed level: */
#define RMS_PERCENTILE 95
/* Duration of RMS window in milliseconds: */
#define RMS_WINDOW_MSECS 50
/* Histogram array elements per dB: */
#define STEPS_PER_DB 100
/* Histogram upper bound in dB (normal max. values in the wild are
* assumed to be around 70, 80 dB): */
#define MAX_DB 120
/* Calibration value: */
#define PINK_REF 64.82 /* 298640883795 */
#define MAX_ORDER MAX (BUTTER_ORDER, YULE_ORDER)
#define MAX_SAMPLE_RATE 48000
/* The + 999 has the effect of ceil()ing: */
#define MAX_SAMPLE_WINDOW (guint) \
((MAX_SAMPLE_RATE * RMS_WINDOW_MSECS + 999) / 1000)
/* Analysis result accumulator. */
struct _RgAnalysisAcc
{
guint32 histogram[STEPS_PER_DB * MAX_DB];
gdouble peak;
};
typedef struct _RgAnalysisAcc RgAnalysisAcc;
/* Analysis context. */
struct _RgAnalysisCtx
{
/* Filter buffers for left channel. */
gfloat inprebuf_l[MAX_ORDER * 2];
gfloat *inpre_l;
gfloat stepbuf_l[MAX_SAMPLE_WINDOW + MAX_ORDER];
gfloat *step_l;
gfloat outbuf_l[MAX_SAMPLE_WINDOW + MAX_ORDER];
gfloat *out_l;
/* Filter buffers for right channel. */
gfloat inprebuf_r[MAX_ORDER * 2];
gfloat *inpre_r;
gfloat stepbuf_r[MAX_SAMPLE_WINDOW + MAX_ORDER];
gfloat *step_r;
gfloat outbuf_r[MAX_SAMPLE_WINDOW + MAX_ORDER];
gfloat *out_r;
/* Number of samples to reach duration of the RMS window: */
guint window_n_samples;
/* Progress of the running window: */
guint window_n_samples_done;
gdouble window_square_sum;
gint sample_rate;
gint sample_rate_index;
RgAnalysisAcc track;
RgAnalysisAcc album;
};
/* Filter coefficients for the IIR filters that form the equal
* loudness filter. XFilter[ctx->sample_rate_index] gives the array
* of the X coefficients (A or B) for the configured sample rate. */
#ifdef G_OS_WIN32
/* Disable double-to-float warning: */
#pragma warning ( disable : 4305 )
#endif
static const gfloat AYule[9][11] = {
{1., -3.84664617118067, 7.81501653005538, -11.34170355132042,
13.05504219327545, -12.28759895145294, 9.48293806319790,
-5.87257861775999, 2.75465861874613, -0.86984376593551,
0.13919314567432},
{1., -3.47845948550071, 6.36317777566148, -8.54751527471874, 9.47693607801280,
-8.81498681370155, 6.85401540936998, -4.39470996079559,
2.19611684890774, -0.75104302451432, 0.13149317958808},
{1., -2.37898834973084, 2.84868151156327, -2.64577170229825, 2.23697657451713,
-1.67148153367602, 1.00595954808547, -0.45953458054983,
0.16378164858596, -0.05032077717131, 0.02347897407020},
{1., -1.61273165137247, 1.07977492259970, -0.25656257754070,
-0.16276719120440, -0.22638893773906, 0.39120800788284,
-0.22138138954925, 0.04500235387352, 0.02005851806501,
0.00302439095741},
{1., -1.49858979367799, 0.87350271418188, 0.12205022308084, -0.80774944671438,
0.47854794562326, -0.12453458140019, -0.04067510197014,
0.08333755284107, -0.04237348025746, 0.02977207319925},
{1., -0.62820619233671, 0.29661783706366, -0.37256372942400, 0.00213767857124,
-0.42029820170918, 0.22199650564824, 0.00613424350682, 0.06747620744683,
0.05784820375801, 0.03222754072173},
{1., -1.04800335126349, 0.29156311971249, -0.26806001042947, 0.00819999645858,
0.45054734505008, -0.33032403314006, 0.06739368333110,
-0.04784254229033, 0.01639907836189, 0.01807364323573},
{1., -0.51035327095184, -0.31863563325245, -0.20256413484477,
0.14728154134330, 0.38952639978999, -0.23313271880868,
-0.05246019024463, -0.02505961724053, 0.02442357316099,
0.01818801111503},
{1., -0.25049871956020, -0.43193942311114, -0.03424681017675,
-0.04678328784242, 0.26408300200955, 0.15113130533216,
-0.17556493366449, -0.18823009262115, 0.05477720428674,
0.04704409688120}
};
static const gfloat BYule[9][11] = {
{0.03857599435200, -0.02160367184185, -0.00123395316851, -0.00009291677959,
-0.01655260341619, 0.02161526843274, -0.02074045215285,
0.00594298065125, 0.00306428023191, 0.00012025322027, 0.00288463683916},
{0.05418656406430, -0.02911007808948, -0.00848709379851, -0.00851165645469,
-0.00834990904936, 0.02245293253339, -0.02596338512915,
0.01624864962975, -0.00240879051584, 0.00674613682247,
-0.00187763777362},
{0.15457299681924, -0.09331049056315, -0.06247880153653, 0.02163541888798,
-0.05588393329856, 0.04781476674921, 0.00222312597743, 0.03174092540049,
-0.01390589421898, 0.00651420667831, -0.00881362733839},
{0.30296907319327, -0.22613988682123, -0.08587323730772, 0.03282930172664,
-0.00915702933434, -0.02364141202522, -0.00584456039913,
0.06276101321749, -0.00000828086748, 0.00205861885564,
-0.02950134983287},
{0.33642304856132, -0.25572241425570, -0.11828570177555, 0.11921148675203,
-0.07834489609479, -0.00469977914380, -0.00589500224440,
0.05724228140351, 0.00832043980773, -0.01635381384540,
-0.01760176568150},
{0.44915256608450, -0.14351757464547, -0.22784394429749, -0.01419140100551,
0.04078262797139, -0.12398163381748, 0.04097565135648, 0.10478503600251,
-0.01863887810927, -0.03193428438915, 0.00541907748707},
{0.56619470757641, -0.75464456939302, 0.16242137742230, 0.16744243493672,
-0.18901604199609, 0.30931782841830, -0.27562961986224,
0.00647310677246, 0.08647503780351, -0.03788984554840,
-0.00588215443421},
{0.58100494960553, -0.53174909058578, -0.14289799034253, 0.17520704835522,
0.02377945217615, 0.15558449135573, -0.25344790059353, 0.01628462406333,
0.06920467763959, -0.03721611395801, -0.00749618797172},
{0.53648789255105, -0.42163034350696, -0.00275953611929, 0.04267842219415,
-0.10214864179676, 0.14590772289388, -0.02459864859345,
-0.11202315195388, -0.04060034127000, 0.04788665548180,
-0.02217936801134}
};
static const gfloat AButter[9][3] = {
{1., -1.97223372919527, 0.97261396931306},
{1., -1.96977855582618, 0.97022847566350},
{1., -1.95835380975398, 0.95920349965459},
{1., -1.95002759149878, 0.95124613669835},
{1., -1.94561023566527, 0.94705070426118},
{1., -1.92783286977036, 0.93034775234268},
{1., -1.91858953033784, 0.92177618768381},
{1., -1.91542108074780, 0.91885558323625},
{1., -1.88903307939452, 0.89487434461664}
};
static const gfloat BButter[9][3] = {
{0.98621192462708, -1.97242384925416, 0.98621192462708},
{0.98500175787242, -1.97000351574484, 0.98500175787242},
{0.97938932735214, -1.95877865470428, 0.97938932735214},
{0.97531843204928, -1.95063686409857, 0.97531843204928},
{0.97316523498161, -1.94633046996323, 0.97316523498161},
{0.96454515552826, -1.92909031105652, 0.96454515552826},
{0.96009142950541, -1.92018285901082, 0.96009142950541},
{0.95856916599601, -1.91713833199203, 0.95856916599601},
{0.94597685600279, -1.89195371200558, 0.94597685600279}
};
#ifdef G_OS_WIN32
#pragma warning ( default : 4305 )
#endif
/* Filter functions. These access elements with negative indices of
* the input and output arrays (up to the filter's order). */
/* For much better performance, the function below has been
* implemented by unrolling the inner loop for our two use cases. */
/*
* static inline void
* apply_filter (const gfloat * input, gfloat * output, guint n_samples,
* const gfloat * a, const gfloat * b, guint order)
* {
* gfloat y;
* gint i, k;
*
* for (i = 0; i < n_samples; i++) {
* y = input[i] * b[0];
* for (k = 1; k <= order; k++)
* y += input[i - k] * b[k] - output[i - k] * a[k];
* output[i] = y;
* }
* }
*/
static inline void
yule_filter (const gfloat * input, gfloat * output,
const gfloat * a, const gfloat * b)
{
output[0] = input[0] * b[0]
+ input[-1] * b[1] - output[-1] * a[1]
+ input[-2] * b[2] - output[-2] * a[2]
+ input[-3] * b[3] - output[-3] * a[3]
+ input[-4] * b[4] - output[-4] * a[4]
+ input[-5] * b[5] - output[-5] * a[5]
+ input[-6] * b[6] - output[-6] * a[6]
+ input[-7] * b[7] - output[-7] * a[7]
+ input[-8] * b[8] - output[-8] * a[8]
+ input[-9] * b[9] - output[-9] * a[9]
+ input[-10] * b[10] - output[-10] * a[10];
}
static inline void
butter_filter (const gfloat * input, gfloat * output,
const gfloat * a, const gfloat * b)
{
output[0] = input[0] * b[0]
+ input[-1] * b[1] - output[-1] * a[1]
+ input[-2] * b[2] - output[-2] * a[2];
}
/* Because butter_filter and yule_filter are inlined, this function is
* a bit blown-up (code-size wise), but not inlining gives a ca. 40%
* performance penalty. */
static inline void
apply_filters (const RgAnalysisCtx * ctx, const gfloat * input_l,
const gfloat * input_r, guint n_samples)
{
const gfloat *ayule = AYule[ctx->sample_rate_index];
const gfloat *byule = BYule[ctx->sample_rate_index];
const gfloat *abutter = AButter[ctx->sample_rate_index];
const gfloat *bbutter = BButter[ctx->sample_rate_index];
gint pos = ctx->window_n_samples_done;
gint i;
for (i = 0; i < n_samples; i++, pos++) {
yule_filter (input_l + i, ctx->step_l + pos, ayule, byule);
butter_filter (ctx->step_l + pos, ctx->out_l + pos, abutter, bbutter);
yule_filter (input_r + i, ctx->step_r + pos, ayule, byule);
butter_filter (ctx->step_r + pos, ctx->out_r + pos, abutter, bbutter);
}
}
/* Clear filter buffer state and current RMS window. */
static void
reset_filters (RgAnalysisCtx * ctx)
{
gint i;
for (i = 0; i < MAX_ORDER; i++) {
ctx->inprebuf_l[i] = 0.;
ctx->stepbuf_l[i] = 0.;
ctx->outbuf_l[i] = 0.;
ctx->inprebuf_r[i] = 0.;
ctx->stepbuf_r[i] = 0.;
ctx->outbuf_r[i] = 0.;
}
ctx->window_square_sum = 0.;
ctx->window_n_samples_done = 0;
}
/* Accumulator functions. */
/* Add two accumulators in-place. The sum is defined as the result of
* the vector sum of the histogram array and the maximum value of the
* peak field. Thus "adding" the accumulators for all tracks yields
* the correct result for obtaining the album gain and peak. */
static void
accumulator_add (RgAnalysisAcc * acc, const RgAnalysisAcc * acc_other)
{
gint i;
for (i = 0; i < G_N_ELEMENTS (acc->histogram); i++)
acc->histogram[i] += acc_other->histogram[i];
acc->peak = MAX (acc->peak, acc_other->peak);
}
/* Reset an accumulator to zero. */
static void
accumulator_clear (RgAnalysisAcc * acc)
{
memset (acc->histogram, 0, sizeof (acc->histogram));
acc->peak = 0.;
}
/* Obtain final analysis result from an accumulator. Returns TRUE on
* success, FALSE on error (if accumulator is still zero). */
static gboolean
accumulator_result (const RgAnalysisAcc * acc, gdouble * result_gain,
gdouble * result_peak)
{
guint32 sum = 0;
guint32 upper;
guint i;
for (i = 0; i < G_N_ELEMENTS (acc->histogram); i++)
sum += acc->histogram[i];
if (sum == 0)
/* All entries are 0: We got less than 50ms of data. */
return FALSE;
upper = (guint32) ceil (sum * (1. - (gdouble) (RMS_PERCENTILE / 100.)));
for (i = G_N_ELEMENTS (acc->histogram); i--;) {
if (upper <= acc->histogram[i])
break;
upper -= acc->histogram[i];
}
if (result_peak != NULL)
*result_peak = acc->peak;
if (result_gain != NULL)
*result_gain = PINK_REF - (gdouble) i / STEPS_PER_DB;
return TRUE;
}
/* Functions that operate on contexts, for external usage. */
/* Create a new context. Before it can be used, a sample rate must be
* configured using rg_analysis_set_sample_rate. */
RgAnalysisCtx *
rg_analysis_new (void)
{
RgAnalysisCtx *ctx;
ctx = g_new (RgAnalysisCtx, 1);
ctx->inpre_l = ctx->inprebuf_l + MAX_ORDER;
ctx->step_l = ctx->stepbuf_l + MAX_ORDER;
ctx->out_l = ctx->outbuf_l + MAX_ORDER;
ctx->inpre_r = ctx->inprebuf_r + MAX_ORDER;
ctx->step_r = ctx->stepbuf_r + MAX_ORDER;
ctx->out_r = ctx->outbuf_r + MAX_ORDER;
ctx->sample_rate = 0;
accumulator_clear (&ctx->track);
accumulator_clear (&ctx->album);
return ctx;
}
/* Adapt to given sample rate. Does nothing if already the current
* rate (returns TRUE then). Returns FALSE only if given sample rate
* is not supported. If the configured rate changes, the last
* unprocessed incomplete 50ms chunk of data is dropped because the
* filters are reset. */
gboolean
rg_analysis_set_sample_rate (RgAnalysisCtx * ctx, gint sample_rate)
{
g_return_val_if_fail (ctx != NULL, FALSE);
if (ctx->sample_rate == sample_rate)
return TRUE;
switch (sample_rate) {
case 48000:
ctx->sample_rate_index = 0;
break;
case 44100:
ctx->sample_rate_index = 1;
break;
case 32000:
ctx->sample_rate_index = 2;
break;
case 24000:
ctx->sample_rate_index = 3;
break;
case 22050:
ctx->sample_rate_index = 4;
break;
case 16000:
ctx->sample_rate_index = 5;
break;
case 12000:
ctx->sample_rate_index = 6;
break;
case 11025:
ctx->sample_rate_index = 7;
break;
case 8000:
ctx->sample_rate_index = 8;
break;
default:
return FALSE;
}
ctx->sample_rate = sample_rate;
/* The + 999 has the effect of ceil()ing: */
ctx->window_n_samples = (guint) ((sample_rate * RMS_WINDOW_MSECS + 999)
/ 1000);
reset_filters (ctx);
return TRUE;
}
void
rg_analysis_destroy (RgAnalysisCtx * ctx)
{
g_free (ctx);
}
/* Entry points for analyzing sample data in common raw data formats.
* The stereo format functions expect interleaved frames. It is
* possible to pass data in different formats for the same context,
* there are no restrictions. All functions have the same signature;
* the depth argument for the float functions is not variable and must
* be given the value 32. */
void
rg_analysis_analyze_mono_float (RgAnalysisCtx * ctx, gconstpointer data,
gsize size, guint depth)
{
gfloat conv_samples[512];
const gfloat *samples = (gfloat *) data;
guint n_samples = size / sizeof (gfloat);
gint i;
g_return_if_fail (depth == 32);
g_return_if_fail (size % sizeof (gfloat) == 0);
while (n_samples) {
gint n = MIN (n_samples, G_N_ELEMENTS (conv_samples));
n_samples -= n;
memcpy (conv_samples, samples, n * sizeof (gfloat));
for (i = 0; i < n; i++) {
ctx->track.peak = MAX (ctx->track.peak, fabs (conv_samples[i]));
conv_samples[i] *= 32768.;
}
samples += n;
rg_analysis_analyze (ctx, conv_samples, NULL, n);
}
}
void
rg_analysis_analyze_stereo_float (RgAnalysisCtx * ctx, gconstpointer data,
gsize size, guint depth)
{
gfloat conv_samples_l[256];
gfloat conv_samples_r[256];
const gfloat *samples = (gfloat *) data;
guint n_frames = size / (sizeof (gfloat) * 2);
gint i;
g_return_if_fail (depth == 32);
g_return_if_fail (size % (sizeof (gfloat) * 2) == 0);
while (n_frames) {
gint n = MIN (n_frames, G_N_ELEMENTS (conv_samples_l));
n_frames -= n;
for (i = 0; i < n; i++) {
gfloat old_sample;
old_sample = samples[2 * i];
ctx->track.peak = MAX (ctx->track.peak, fabs (old_sample));
conv_samples_l[i] = old_sample * 32768.;
old_sample = samples[2 * i + 1];
ctx->track.peak = MAX (ctx->track.peak, fabs (old_sample));
conv_samples_r[i] = old_sample * 32768.;
}
samples += 2 * n;
rg_analysis_analyze (ctx, conv_samples_l, conv_samples_r, n);
}
}
void
rg_analysis_analyze_mono_int16 (RgAnalysisCtx * ctx, gconstpointer data,
gsize size, guint depth)
{
gfloat conv_samples[512];
gint32 peak_sample = 0;
const gint16 *samples = (gint16 *) data;
guint n_samples = size / sizeof (gint16);
gint shift = sizeof (gint16) * 8 - depth;
gint i;
g_return_if_fail (depth <= (sizeof (gint16) * 8));
g_return_if_fail (size % sizeof (gint16) == 0);
while (n_samples) {
gint n = MIN (n_samples, G_N_ELEMENTS (conv_samples));
n_samples -= n;
for (i = 0; i < n; i++) {
gint16 old_sample = samples[i] << shift;
peak_sample = MAX (peak_sample, ABS ((gint32) old_sample));
conv_samples[i] = (gfloat) old_sample;
}
samples += n;
rg_analysis_analyze (ctx, conv_samples, NULL, n);
}
ctx->track.peak = MAX (ctx->track.peak,
(gdouble) peak_sample / ((gdouble) (1u << 15)));
}
void
rg_analysis_analyze_stereo_int16 (RgAnalysisCtx * ctx, gconstpointer data,
gsize size, guint depth)
{
gfloat conv_samples_l[256];
gfloat conv_samples_r[256];
gint32 peak_sample = 0;
const gint16 *samples = (gint16 *) data;
guint n_frames = size / (sizeof (gint16) * 2);
gint shift = sizeof (gint16) * 8 - depth;
gint i;
g_return_if_fail (depth <= (sizeof (gint16) * 8));
g_return_if_fail (size % (sizeof (gint16) * 2) == 0);
while (n_frames) {
gint n = MIN (n_frames, G_N_ELEMENTS (conv_samples_l));
n_frames -= n;
for (i = 0; i < n; i++) {
gint16 old_sample;
old_sample = samples[2 * i] << shift;
peak_sample = MAX (peak_sample, ABS ((gint32) old_sample));
conv_samples_l[i] = (gfloat) old_sample;
old_sample = samples[2 * i + 1] << shift;
peak_sample = MAX (peak_sample, ABS ((gint32) old_sample));
conv_samples_r[i] = (gfloat) old_sample;
}
samples += 2 * n;
rg_analysis_analyze (ctx, conv_samples_l, conv_samples_r, n);
}
ctx->track.peak = MAX (ctx->track.peak,
(gdouble) peak_sample / ((gdouble) (1u << 15)));
}
/* Analyze the given chunk of samples. The sample data is given in
* floating point format but should be scaled such that the values
* +/-32768.0 correspond to the -0dBFS reference amplitude.
*
* samples_l: Buffer with sample data for the left channel or of the
* mono channel.
*
* samples_r: Buffer with sample data for the right channel or NULL
* for mono.
*
* n_samples: Number of samples passed in each buffer.
*/
void
rg_analysis_analyze (RgAnalysisCtx * ctx, const gfloat * samples_l,
const gfloat * samples_r, guint n_samples)
{
const gfloat *input_l, *input_r;
guint n_samples_done;
gint i;
g_return_if_fail (ctx != NULL);
g_return_if_fail (samples_l != NULL);
g_return_if_fail (ctx->sample_rate != 0);
if (n_samples == 0)
return;
if (samples_r == NULL)
/* Mono. */
samples_r = samples_l;
memcpy (ctx->inpre_l, samples_l,
MIN (n_samples, MAX_ORDER) * sizeof (gfloat));
memcpy (ctx->inpre_r, samples_r,
MIN (n_samples, MAX_ORDER) * sizeof (gfloat));
n_samples_done = 0;
while (n_samples_done < n_samples) {
/* Limit number of samples to be processed in this iteration to
* the number needed to complete the next window: */
guint n_samples_current = MIN (n_samples - n_samples_done,
ctx->window_n_samples - ctx->window_n_samples_done);
if (n_samples_done < MAX_ORDER) {
input_l = ctx->inpre_l + n_samples_done;
input_r = ctx->inpre_r + n_samples_done;
n_samples_current = MIN (n_samples_current, MAX_ORDER - n_samples_done);
} else {
input_l = samples_l + n_samples_done;
input_r = samples_r + n_samples_done;
}
apply_filters (ctx, input_l, input_r, n_samples_current);
/* Update the square sum. */
for (i = 0; i < n_samples_current; i++)
ctx->window_square_sum += ctx->out_l[ctx->window_n_samples_done + i]
* ctx->out_l[ctx->window_n_samples_done + i]
+ ctx->out_r[ctx->window_n_samples_done + i]
* ctx->out_r[ctx->window_n_samples_done + i];
ctx->window_n_samples_done += n_samples_current;
g_return_if_fail (ctx->window_n_samples_done <= ctx->window_n_samples);
if (ctx->window_n_samples_done == ctx->window_n_samples) {
/* Get the Root Mean Square (RMS) for this set of samples. */
gdouble val = STEPS_PER_DB * 10. * log10 (ctx->window_square_sum /
ctx->window_n_samples * 0.5 + 1.e-37);
gint ival = CLAMP ((gint) val, 0,
(gint) G_N_ELEMENTS (ctx->track.histogram) - 1);
ctx->track.histogram[ival]++;
ctx->window_square_sum = 0.;
ctx->window_n_samples_done = 0;
/* No need for memmove here, the areas never overlap: Even for
* the smallest sample rate, the number of samples needed for
* the window is greater than MAX_ORDER. */
memcpy (ctx->stepbuf_l, ctx->stepbuf_l + ctx->window_n_samples,
MAX_ORDER * sizeof (gfloat));
memcpy (ctx->outbuf_l, ctx->outbuf_l + ctx->window_n_samples,
MAX_ORDER * sizeof (gfloat));
memcpy (ctx->stepbuf_r, ctx->stepbuf_r + ctx->window_n_samples,
MAX_ORDER * sizeof (gfloat));
memcpy (ctx->outbuf_r, ctx->outbuf_r + ctx->window_n_samples,
MAX_ORDER * sizeof (gfloat));
}
n_samples_done += n_samples_current;
}
if (n_samples >= MAX_ORDER) {
memcpy (ctx->inprebuf_l, samples_l + n_samples - MAX_ORDER,
MAX_ORDER * sizeof (gfloat));
memcpy (ctx->inprebuf_r, samples_r + n_samples - MAX_ORDER,
MAX_ORDER * sizeof (gfloat));
} else {
memmove (ctx->inprebuf_l, ctx->inprebuf_l + n_samples,
(MAX_ORDER - n_samples) * sizeof (gfloat));
memcpy (ctx->inprebuf_l + MAX_ORDER - n_samples, samples_l,
n_samples * sizeof (gfloat));
memmove (ctx->inprebuf_r, ctx->inprebuf_r + n_samples,
(MAX_ORDER - n_samples) * sizeof (gfloat));
memcpy (ctx->inprebuf_r + MAX_ORDER - n_samples, samples_r,
n_samples * sizeof (gfloat));
}
}
/* Obtain track gain and peak. Returns TRUE on success. Can fail if
* not enough samples have been processed. Updates album accumulator.
* Resets track accumulator. */
gboolean
rg_analysis_track_result (RgAnalysisCtx * ctx, gdouble * gain, gdouble * peak)
{
gboolean result;
g_return_val_if_fail (ctx != NULL, FALSE);
accumulator_add (&ctx->album, &ctx->track);
result = accumulator_result (&ctx->track, gain, peak);
accumulator_clear (&ctx->track);
reset_filters (ctx);
return result;
}
/* Obtain album gain and peak. Returns TRUE on success. Can fail if
* not enough samples have been processed. Resets album
* accumulator. */
gboolean
rg_analysis_album_result (RgAnalysisCtx * ctx, gdouble * gain, gdouble * peak)
{
gboolean result;
g_return_val_if_fail (ctx != NULL, FALSE);
result = accumulator_result (&ctx->album, gain, peak);
accumulator_clear (&ctx->album);
return result;
}
void
rg_analysis_reset_album (RgAnalysisCtx * ctx)
{
accumulator_clear (&ctx->album);
}
/* Reset internal buffers as well as track and album accumulators.
* Configured sample rate is kept intact. */
void
rg_analysis_reset (RgAnalysisCtx * ctx)
{
g_return_if_fail (ctx != NULL);
reset_filters (ctx);
accumulator_clear (&ctx->track);
accumulator_clear (&ctx->album);
}

View file

@ -0,0 +1,58 @@
/* GStreamer ReplayGain analysis
*
* Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
* Copyright (C) 2001 David Robinson <David@Robinson.org>
* Glen Sawyer <glensawyer@hotmail.com>
*
* rganalysis.h: Analyze raw audio data in accordance with ReplayGain
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public License
* as published by the Free Software Foundation; either version 2.1 of
* the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful, but
* WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with this library; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
* 02110-1301 USA
*/
#ifndef __RG_ANALYSIS_H__
#define __RG_ANALYSIS_H__
#include <glib.h>
G_BEGIN_DECLS
#define RG_REFERENCE_LEVEL 89.
typedef struct _RgAnalysisCtx RgAnalysisCtx;
RgAnalysisCtx *rg_analysis_new (void);
gboolean rg_analysis_set_sample_rate (RgAnalysisCtx * ctx, gint sample_rate);
void rg_analysis_analyze_mono_float (RgAnalysisCtx * ctx, gconstpointer data,
gsize size, guint depth);
void rg_analysis_analyze_stereo_float (RgAnalysisCtx * ctx, gconstpointer data,
gsize size, guint depth);
void rg_analysis_analyze_mono_int16 (RgAnalysisCtx * ctx, gconstpointer data,
gsize size, guint depth);
void rg_analysis_analyze_stereo_int16 (RgAnalysisCtx * ctx, gconstpointer data,
gsize size, guint depth);
void rg_analysis_analyze (RgAnalysisCtx * ctx, const gfloat * samples_l,
const gfloat * samples_r, guint n_samples);
gboolean rg_analysis_track_result (RgAnalysisCtx * ctx, gdouble * gain,
gdouble * peak);
gboolean rg_analysis_album_result (RgAnalysisCtx * ctx, gdouble * gain,
gdouble * peak);
void rg_analysis_reset_album (RgAnalysisCtx * ctx);
void rg_analysis_reset (RgAnalysisCtx * ctx);
void rg_analysis_destroy (RgAnalysisCtx * ctx);
G_END_DECLS
#endif /* __RG_ANALYSIS_H__ */

View file

@ -52,6 +52,7 @@ VALGRIND_TESTS_DISABLE = \
check_PROGRAMS = \ check_PROGRAMS = \
$(check_mpeg2enc) \ $(check_mpeg2enc) \
elements/rganalysis \
elements/videocrop \ elements/videocrop \
$(check_wavpack) $(check_wavpack)

View file

@ -8,3 +8,4 @@ wavpackdec
wavpackenc wavpackenc
wavpackparse wavpackparse
videocrop videocrop
rganalysis

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