From e985585a4ec8ec1a681c9643f6727a230fb536d7 Mon Sep 17 00:00:00 2001 From: Jan Schmidt Date: Sat, 19 Jul 2008 00:58:49 +0000 Subject: Remove interleave and replaygain plugins that have moved to -good Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-replaygain.xml: * gst/interleave/Makefile.am: * gst/interleave/deinterleave.c: * gst/interleave/deinterleave.h: * gst/interleave/interleave.c: * gst/interleave/interleave.h: * gst/interleave/plugin.c: * gst/interleave/plugin.h: * gst/replaygain/Makefile.am: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrganalysis.h: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrglimiter.h: * gst/replaygain/gstrgvolume.c: * gst/replaygain/gstrgvolume.h: * gst/replaygain/replaygain.c: * gst/replaygain/replaygain.h: * gst/replaygain/rganalysis.c: * gst/replaygain/rganalysis.h: * tests/check/Makefile.am: * tests/check/elements/deinterleave.c: * tests/check/elements/interleave.c: * tests/check/elements/rganalysis.c: * tests/check/elements/rglimiter.c: * tests/check/elements/rgvolume.c: Remove interleave and replaygain plugins that have moved to -good --- gst/replaygain/gstrganalysis.c | 692 ----------------------------------------- 1 file changed, 692 deletions(-) delete mode 100644 gst/replaygain/gstrganalysis.c (limited to 'gst/replaygain/gstrganalysis.c') diff --git a/gst/replaygain/gstrganalysis.c b/gst/replaygain/gstrganalysis.c deleted file mode 100644 index 982c8a7f..00000000 --- a/gst/replaygain/gstrganalysis.c +++ /dev/null @@ -1,692 +0,0 @@ -/* GStreamer ReplayGain analysis - * - * Copyright (C) 2006 Rene Stadler - * - * 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 - * @see_also: #GstRgVolume - * - * This element analyzes raw audio sample data in accordance with the proposed - * ReplayGain standard for - * calculating the ideal replay gain for music tracks and albums. The element - * is designed as a pass-through filter that never modifies any data. As it - * receives an EOS event, it finalizes the ongoing analysis and generates a tag - * list containing the results. It is sent downstream with a tag event and - * posted on the message bus with a tag message. The EOS event is forwarded as - * normal afterwards. Result tag lists at least contain the tags - * #GST_TAG_TRACK_GAIN, #GST_TAG_TRACK_PEAK and #GST_TAG_REFERENCE_LEVEL. - * - * 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. - * - * - * Example launch lines - * |[ - * gst-launch -t audiotestsrc wave=sine num-buffers=512 ! rganalysis ! fakesink - * ]| Analyze a simple test waveform - * |[ - * gst-launch -t filesrc location=filename.ext ! decodebin \ - * ! audioconvert ! audioresample ! rganalysis ! fakesink - * ]| Analyze a given file - * |[ - * gst-launch -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav \ - * ! wavparse ! rganalysis ! fakesink - * ]| Analyze the pink noise reference file - * - * The above launch line yields a result gain of +6 dB (instead of the expected - * +0 dB). This is not in error, refer to the #GstRgAnalysis:reference-level - * property documentation for more information. - * - * - * - * Acknowledgements - * - * This element is based on code used in the vorbisgain program and many - * others. The relevant parts are copyrighted by David Robinson, Glen Sawyer - * and Frank Klemm. - * - * - */ - -#ifdef HAVE_CONFIG_H -#include -#endif - -#include -#include - -#include "gstrganalysis.h" -#include "replaygain.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 " -}; - -/* 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; - - /** - * GstRgAnalysis:num-tracks: - * - * Number of remaining album tracks. - * - * Analyzing several streams sequentially and assigning them a common result - * gain is known as "album processing". If this gain is used during playback - * (by switching to "album mode"), all tracks of an album receive the same - * amplification. This keeps the relative volume levels between the tracks - * intact. To enable this, set this property to the number of streams that - * will be processed as album tracks. - * - * Every time an EOS event is received, the value of this property is - * decremented by one. As it reaches zero, it is assumed that the last track - * of the album finished. The tag list for the final stream will contain the - * additional tags #GST_TAG_ALBUM_GAIN and #GST_TAG_ALBUM_PEAK. All other - * streams just get the two track tags posted because the values for the album - * tags are not known before all tracks are analyzed. Applications need to - * ensure that the album gain and peak values are also associated with the - * other tracks when storing the results. - * - * If the total number of album tracks is unknown beforehand, just ensure that - * the value is greater than 1 before each track starts. Then before the end - * of the last track, set it to the value 1. - * - * To perform album processing, the element has to preserve data between - * streams. This cannot survive a state change to the NULL or READY state. - * If you change your pipeline's state to NULL or READY between tracks, lock - * the element's state using gst_element_set_locked_state() when it is in - * PAUSED or PLAYING. - */ - g_object_class_install_property (gobject_class, PROP_NUM_TRACKS, - g_param_spec_int ("num-tracks", "Number of album tracks", - "Number of remaining album tracks", 0, G_MAXINT, 0, - G_PARAM_READWRITE)); - /** - * GstRgAnalysis:forced: - * - * Whether to analyze streams even when ReplayGain tags exist. - * - * For assisting transcoder/converter applications, the element can silently - * skip the processing of streams that already contain the necessary tags. - * Data will flow as usual but the element will not consume CPU time and will - * not generate result tags. To enable possible skipping, set this property - * to #FALSE. - * - * If used in conjunction with 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, the application has to scan all files beforehand and enable - * forcing of processing if needed. - */ - g_object_class_install_property (gobject_class, PROP_FORCED, - g_param_spec_boolean ("forced", "Forced", - "Analyze even if ReplayGain tags exist", - FORCED_DEFAULT, G_PARAM_READWRITE)); - /** - * GstRgAnalysis:reference-level: - * - * Reference level [dB]. - * - * Analyzing the ReplayGain pink noise reference waveform computes a result of - * +6 dB instead of the expected 0 dB. This is because the default reference - * level is 89 dB. To obtain values as lined out in the original proposal of - * ReplayGain, set this property to 83. - * - * Almost all software uses 89 dB as a reference however, and this value has - * become the new official value. That is to say, while the change has been - * acclaimed by the author of the ReplayGain proposal, the webpage is still outdated at the time - * of this writing. - * - * The value was changed because the original proposal recommends a default - * pre-amp value of +6 dB for playback. This seemed a bit odd, as it means - * that the algorithm has the general tendency to produce adjustment values - * that are 6 dB too low. Bumping the reference level by 6 dB compensated for - * this. - * - * The problem of the reference level being ambiguous for lack of concise - * standardization is to be solved by adopting the #GST_TAG_REFERENCE_LEVEL - * tag, which allows to store the used value alongside the gain values. - */ - g_object_class_install_property (gobject_class, PROP_REFERENCE_LEVEL, - g_param_spec_double ("reference-level", "Reference level", - "Reference level [dB]", 0.0, 150., 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) -{ - GstBaseTransform *base = GST_BASE_TRANSFORM (filter); - - gst_base_transform_set_gap_aware (base, TRUE); - - 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_LOG_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 *name; - 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); - name = 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 (g_str_equal (name, "audio/x-raw-float")) { - - 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 (g_str_equal (name, "audio/x-raw-int")) { - - 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; - - /* Buffers made up of silence have no influence on the analysis: */ - if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)) - return GST_FLOW_OK; - - GST_LOG_OBJECT (filter, "processing buffer of size %u", - GST_BUFFER_SIZE (buf)); - - filter->analyze (filter->ctx, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), - 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; -} -- cgit v1.2.1