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723 lines
24 KiB
C
723 lines
24 KiB
C
/* GStreamer ReplayGain analysis
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*
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* Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
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*
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* gstrganalysis.c: Element that performs the ReplayGain analysis
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public License
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* as published by the Free Software Foundation; either version 2.1 of
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* the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful, but
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* WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with this library; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
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* 02110-1301 USA
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*/
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/**
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* SECTION:element-rganalysis
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* @see_also: #GstRgVolume
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*
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* This element analyzes raw audio sample data in accordance with the proposed
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* <ulink url="http://replaygain.org">ReplayGain standard</ulink> for
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* calculating the ideal replay gain for music tracks and albums. The element
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* is designed as a pass-through filter that never modifies any data. As it
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* receives an EOS event, it finalizes the ongoing analysis and generates a tag
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* list containing the results. It is sent downstream with a tag event and
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* posted on the message bus with a tag message. The EOS event is forwarded as
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* normal afterwards. Result tag lists at least contain the tags
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* #GST_TAG_TRACK_GAIN, #GST_TAG_TRACK_PEAK and #GST_TAG_REFERENCE_LEVEL.
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*
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* Because the generated metadata tags become available at the end of streams,
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* downstream muxer and encoder elements are normally unable to save them in
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* their output since they generally save metadata in the file header.
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* Therefore, it is often necessary that applications read the results in a bus
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* event handler for the tag message. Obtaining the values this way is always
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* needed for <link linkend="GstRgAnalysis--num-tracks">album processing</link>
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* since the album gain and peak values need to be associated with all tracks of
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* an album, not just the last one.
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*
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* <refsect2>
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* <title>Example launch lines</title>
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* |[
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* gst-launch -t audiotestsrc wave=sine num-buffers=512 ! rganalysis ! fakesink
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* ]| Analyze a simple test waveform
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* |[
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* gst-launch -t filesrc location=filename.ext ! decodebin \
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* ! audioconvert ! audioresample ! rganalysis ! fakesink
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* ]| Analyze a given file
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* |[
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* gst-launch -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav \
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* ! wavparse ! rganalysis ! fakesink
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* ]| Analyze the pink noise reference file
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* <para>
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* The above launch line yields a result gain of +6 dB (instead of the expected
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* +0 dB). This is not in error, refer to the #GstRgAnalysis:reference-level
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* property documentation for more information.
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* </para>
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* </refsect2>
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* <refsect2>
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* <title>Acknowledgements</title>
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* <para>
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* This element is based on code used in the <ulink
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* url="http://sjeng.org/vorbisgain.html">vorbisgain</ulink> program and many
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* others. The relevant parts are copyrighted by David Robinson, Glen Sawyer
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* and Frank Klemm.
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* </para>
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* </refsect2>
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*/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include <gst/gst.h>
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#include <gst/base/gstbasetransform.h>
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#include "gstrganalysis.h"
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#include "replaygain.h"
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GST_DEBUG_CATEGORY_STATIC (gst_rg_analysis_debug);
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#define GST_CAT_DEFAULT gst_rg_analysis_debug
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/* Default property value. */
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#define FORCED_DEFAULT TRUE
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#define DEFAULT_MESSAGE FALSE
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enum
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{
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PROP_0,
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PROP_NUM_TRACKS,
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PROP_FORCED,
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PROP_REFERENCE_LEVEL,
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PROP_MESSAGE
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};
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/* The ReplayGain algorithm is intended for use with mono and stereo
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* audio. The used implementation has filter coefficients for the
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* "usual" sample rates in the 8000 to 48000 Hz range. */
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#define REPLAY_GAIN_CAPS \
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"channels = (int) { 1, 2 }, " \
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"rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, " \
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"44100, 48000 }"
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static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
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GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
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"width = (int) 32, " "endianness = (int) BYTE_ORDER, "
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REPLAY_GAIN_CAPS "; "
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"audio/x-raw-int, "
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"width = (int) 16, " "depth = (int) [ 1, 16 ], "
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"signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
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REPLAY_GAIN_CAPS));
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static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
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GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
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"width = (int) 32, " "endianness = (int) BYTE_ORDER, "
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REPLAY_GAIN_CAPS "; "
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"audio/x-raw-int, "
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"width = (int) 16, " "depth = (int) [ 1, 16 ], "
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"signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
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REPLAY_GAIN_CAPS));
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GST_BOILERPLATE (GstRgAnalysis, gst_rg_analysis, GstBaseTransform,
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GST_TYPE_BASE_TRANSFORM);
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static void gst_rg_analysis_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec);
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static void gst_rg_analysis_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec);
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static gboolean gst_rg_analysis_start (GstBaseTransform * base);
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static gboolean gst_rg_analysis_set_caps (GstBaseTransform * base,
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GstCaps * incaps, GstCaps * outcaps);
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static GstFlowReturn gst_rg_analysis_transform_ip (GstBaseTransform * base,
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GstBuffer * buf);
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static gboolean gst_rg_analysis_event (GstBaseTransform * base,
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GstEvent * event);
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static gboolean gst_rg_analysis_stop (GstBaseTransform * base);
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static void gst_rg_analysis_handle_tags (GstRgAnalysis * filter,
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const GstTagList * tag_list);
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static void gst_rg_analysis_handle_eos (GstRgAnalysis * filter);
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static gboolean gst_rg_analysis_track_result (GstRgAnalysis * filter,
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GstTagList ** tag_list);
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static gboolean gst_rg_analysis_album_result (GstRgAnalysis * filter,
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GstTagList ** tag_list);
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static void
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gst_rg_analysis_base_init (gpointer g_class)
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{
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GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&src_factory));
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gst_element_class_add_pad_template (element_class,
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gst_static_pad_template_get (&sink_factory));
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gst_element_class_set_details_simple (element_class, "ReplayGain analysis",
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"Filter/Analyzer/Audio",
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"Perform the ReplayGain analysis",
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"Ren\xc3\xa9 Stadler <mail@renestadler.de>");
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GST_DEBUG_CATEGORY_INIT (gst_rg_analysis_debug, "rganalysis", 0,
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"ReplayGain analysis element");
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}
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static void
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gst_rg_analysis_class_init (GstRgAnalysisClass * klass)
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{
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GObjectClass *gobject_class;
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GstBaseTransformClass *trans_class;
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gobject_class = (GObjectClass *) klass;
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gobject_class->set_property = gst_rg_analysis_set_property;
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gobject_class->get_property = gst_rg_analysis_get_property;
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/**
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* GstRgAnalysis:num-tracks:
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*
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* Number of remaining album tracks.
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*
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* Analyzing several streams sequentially and assigning them a common result
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* gain is known as "album processing". If this gain is used during playback
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* (by switching to "album mode"), all tracks of an album receive the same
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* amplification. This keeps the relative volume levels between the tracks
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* intact. To enable this, set this property to the number of streams that
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* will be processed as album tracks.
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*
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* Every time an EOS event is received, the value of this property is
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* decremented by one. As it reaches zero, it is assumed that the last track
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* of the album finished. The tag list for the final stream will contain the
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* additional tags #GST_TAG_ALBUM_GAIN and #GST_TAG_ALBUM_PEAK. All other
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* streams just get the two track tags posted because the values for the album
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* tags are not known before all tracks are analyzed. Applications need to
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* ensure that the album gain and peak values are also associated with the
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* other tracks when storing the results.
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*
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* If the total number of album tracks is unknown beforehand, just ensure that
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* the value is greater than 1 before each track starts. Then before the end
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* of the last track, set it to the value 1.
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*
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* To perform album processing, the element has to preserve data between
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* streams. This cannot survive a state change to the NULL or READY state.
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* If you change your pipeline's state to NULL or READY between tracks, lock
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* the element's state using gst_element_set_locked_state() when it is in
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* PAUSED or PLAYING.
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*/
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g_object_class_install_property (gobject_class, PROP_NUM_TRACKS,
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g_param_spec_int ("num-tracks", "Number of album tracks",
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"Number of remaining album tracks", 0, G_MAXINT, 0,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRgAnalysis:forced:
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*
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* Whether to analyze streams even when ReplayGain tags exist.
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*
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* For assisting transcoder/converter applications, the element can silently
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* skip the processing of streams that already contain the necessary tags.
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* Data will flow as usual but the element will not consume CPU time and will
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* not generate result tags. To enable possible skipping, set this property
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* to #FALSE.
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*
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* If used in conjunction with <link linkend="GstRgAnalysis--num-tracks">album
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* processing</link>, the element will skip the number of remaining album
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* tracks if a full set of tags is found for the first track. If a subsequent
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* track of the album is missing tags, processing cannot start again. If this
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* is undesired, the application has to scan all files beforehand and enable
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* forcing of processing if needed.
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*/
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g_object_class_install_property (gobject_class, PROP_FORCED,
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g_param_spec_boolean ("forced", "Forced",
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"Analyze even if ReplayGain tags exist",
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FORCED_DEFAULT, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRgAnalysis:reference-level:
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*
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* Reference level [dB].
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*
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* Analyzing the ReplayGain pink noise reference waveform computes a result of
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* +6 dB instead of the expected 0 dB. This is because the default reference
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* level is 89 dB. To obtain values as lined out in the original proposal of
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* ReplayGain, set this property to 83.
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*
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* Almost all software uses 89 dB as a reference however, and this value has
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* become the new official value. That is to say, while the change has been
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* acclaimed by the author of the ReplayGain proposal, the <ulink
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* url="http://replaygain.org">webpage</ulink> is still outdated at the time
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* of this writing.
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*
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* The value was changed because the original proposal recommends a default
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* pre-amp value of +6 dB for playback. This seemed a bit odd, as it means
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* that the algorithm has the general tendency to produce adjustment values
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* that are 6 dB too low. Bumping the reference level by 6 dB compensated for
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* this.
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*
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* The problem of the reference level being ambiguous for lack of concise
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* standardization is to be solved by adopting the #GST_TAG_REFERENCE_LEVEL
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* tag, which allows to store the used value alongside the gain values.
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*/
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g_object_class_install_property (gobject_class, PROP_REFERENCE_LEVEL,
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g_param_spec_double ("reference-level", "Reference level",
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"Reference level [dB]", 0.0, 150., RG_REFERENCE_LEVEL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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g_object_class_install_property (G_OBJECT_CLASS (klass), PROP_MESSAGE,
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g_param_spec_boolean ("message", "Message",
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"Post statics messages",
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DEFAULT_MESSAGE,
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G_PARAM_READWRITE | G_PARAM_CONSTRUCT | G_PARAM_STATIC_STRINGS));
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trans_class = (GstBaseTransformClass *) klass;
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trans_class->start = GST_DEBUG_FUNCPTR (gst_rg_analysis_start);
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trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_rg_analysis_set_caps);
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trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_analysis_transform_ip);
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trans_class->event = GST_DEBUG_FUNCPTR (gst_rg_analysis_event);
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trans_class->stop = GST_DEBUG_FUNCPTR (gst_rg_analysis_stop);
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trans_class->passthrough_on_same_caps = TRUE;
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}
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static void
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gst_rg_analysis_init (GstRgAnalysis * filter, GstRgAnalysisClass * gclass)
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{
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GstBaseTransform *base = GST_BASE_TRANSFORM (filter);
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gst_base_transform_set_gap_aware (base, TRUE);
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filter->num_tracks = 0;
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filter->forced = FORCED_DEFAULT;
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filter->message = DEFAULT_MESSAGE;
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filter->reference_level = RG_REFERENCE_LEVEL;
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filter->ctx = NULL;
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filter->analyze = NULL;
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}
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static void
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gst_rg_analysis_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRgAnalysis *filter = GST_RG_ANALYSIS (object);
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GST_OBJECT_LOCK (filter);
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switch (prop_id) {
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case PROP_NUM_TRACKS:
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filter->num_tracks = g_value_get_int (value);
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break;
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case PROP_FORCED:
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filter->forced = g_value_get_boolean (value);
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break;
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case PROP_REFERENCE_LEVEL:
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filter->reference_level = g_value_get_double (value);
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break;
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case PROP_MESSAGE:
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filter->message = g_value_get_boolean (value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_OBJECT_UNLOCK (filter);
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}
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static void
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gst_rg_analysis_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRgAnalysis *filter = GST_RG_ANALYSIS (object);
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GST_OBJECT_LOCK (filter);
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switch (prop_id) {
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case PROP_NUM_TRACKS:
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g_value_set_int (value, filter->num_tracks);
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break;
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case PROP_FORCED:
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g_value_set_boolean (value, filter->forced);
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break;
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case PROP_REFERENCE_LEVEL:
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g_value_set_double (value, filter->reference_level);
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break;
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case PROP_MESSAGE:
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g_value_set_boolean (value, filter->message);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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GST_OBJECT_UNLOCK (filter);
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}
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static void
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gst_rg_analysis_post_message (gpointer rganalysis, GstClockTime timestamp,
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GstClockTime duration, gdouble rglevel)
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{
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GstRgAnalysis *filter = GST_RG_ANALYSIS (rganalysis);
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if (filter->message) {
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GstMessage *m;
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m = gst_message_new_element (GST_OBJECT_CAST (rganalysis),
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gst_structure_new ("rganalysis",
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"timestamp", G_TYPE_UINT64, timestamp,
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"duration", G_TYPE_UINT64, duration,
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"rglevel", G_TYPE_DOUBLE, rglevel, NULL));
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gst_element_post_message (GST_ELEMENT_CAST (rganalysis), m);
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}
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}
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static gboolean
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gst_rg_analysis_start (GstBaseTransform * base)
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{
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GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
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filter->ignore_tags = FALSE;
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filter->skip = FALSE;
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filter->has_track_gain = FALSE;
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filter->has_track_peak = FALSE;
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filter->has_album_gain = FALSE;
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filter->has_album_peak = FALSE;
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filter->ctx = rg_analysis_new ();
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GST_OBJECT_LOCK (filter);
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rg_analysis_init_silence_detection (filter->ctx, gst_rg_analysis_post_message,
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filter);
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GST_OBJECT_UNLOCK (filter);
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filter->analyze = NULL;
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GST_LOG_OBJECT (filter, "started");
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return TRUE;
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}
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static gboolean
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gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
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GstCaps * out_caps)
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{
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GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
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GstStructure *structure;
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const gchar *name;
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gint n_channels, sample_rate, sample_bit_size, sample_size;
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g_return_val_if_fail (filter->ctx != NULL, FALSE);
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GST_DEBUG_OBJECT (filter,
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"set_caps in %" GST_PTR_FORMAT " out %" GST_PTR_FORMAT,
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in_caps, out_caps);
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structure = gst_caps_get_structure (in_caps, 0);
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name = gst_structure_get_name (structure);
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if (!gst_structure_get_int (structure, "width", &sample_bit_size)
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|| !gst_structure_get_int (structure, "channels", &n_channels)
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|| !gst_structure_get_int (structure, "rate", &sample_rate))
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goto invalid_format;
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if (!rg_analysis_set_sample_rate (filter->ctx, sample_rate))
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goto invalid_format;
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if (sample_bit_size % 8 != 0)
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goto invalid_format;
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sample_size = sample_bit_size / 8;
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if (g_str_equal (name, "audio/x-raw-float")) {
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if (sample_size != sizeof (gfloat))
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goto invalid_format;
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/* The depth is not variable for float formats of course. It just
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* makes the transform function nice and simple if the
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* rg_analysis_analyze_* functions have a common signature. */
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filter->depth = sizeof (gfloat) * 8;
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if (n_channels == 1)
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filter->analyze = rg_analysis_analyze_mono_float;
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else if (n_channels == 2)
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filter->analyze = rg_analysis_analyze_stereo_float;
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else
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goto invalid_format;
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} else if (g_str_equal (name, "audio/x-raw-int")) {
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|
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_WRONG_STATE);
|
|
g_return_val_if_fail (filter->analyze != NULL, GST_FLOW_NOT_NEGOTIATED);
|
|
|
|
if (filter->skip)
|
|
return GST_FLOW_OK;
|
|
|
|
GST_LOG_OBJECT (filter, "processing buffer of size %u",
|
|
GST_BUFFER_SIZE (buf));
|
|
|
|
rg_analysis_start_buffer (filter->ctx, GST_BUFFER_TIMESTAMP (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_LOG_OBJECT (filter, "received EOS event");
|
|
|
|
gst_rg_analysis_handle_eos (filter);
|
|
|
|
GST_LOG_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 GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
|
|
}
|
|
|
|
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_LOG_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_DEBUG_OBJECT (filter, "ignoring tag event: skipping album");
|
|
return;
|
|
} else if (filter->skip) {
|
|
GST_DEBUG_OBJECT (filter, "ignoring tag event: skipping track");
|
|
return;
|
|
} else if (filter->ignore_tags) {
|
|
GST_DEBUG_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_DEBUG_OBJECT (filter, "track tags not complete yet");
|
|
return;
|
|
}
|
|
|
|
if (album_processing && !(filter->has_album_gain && filter->has_album_peak)) {
|
|
GST_DEBUG_OBJECT (filter, "album tags not complete yet");
|
|
return;
|
|
}
|
|
|
|
if (filter->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_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
|
|
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_LOG_OBJECT (filter, "posting tag list with results");
|
|
gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
|
|
GST_TAG_REFERENCE_LEVEL, filter->reference_level, NULL);
|
|
/* This steals our reference to the list: */
|
|
gst_element_found_tags_for_pad (GST_ELEMENT (filter),
|
|
GST_BASE_TRANSFORM_SRC_PAD (GST_BASE_TRANSFORM (filter)), tag_list);
|
|
}
|
|
}
|
|
|
|
if (album_processing) {
|
|
filter->num_tracks--;
|
|
|
|
if (!album_finished) {
|
|
GST_DEBUG_OBJECT (filter, "album not finished yet (num-tracks is now %u)",
|
|
filter->num_tracks);
|
|
} else {
|
|
GST_DEBUG_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;
|
|
}
|