diff options
Diffstat (limited to 'gst/replaygain')
| -rw-r--r-- | gst/replaygain/Makefile.am | 21 | ||||
| -rw-r--r-- | gst/replaygain/gstrganalysis.c | 692 | ||||
| -rw-r--r-- | gst/replaygain/gstrganalysis.h | 85 | ||||
| -rw-r--r-- | gst/replaygain/gstrglimiter.c | 202 | ||||
| -rw-r--r-- | gst/replaygain/gstrglimiter.h | 64 | ||||
| -rw-r--r-- | gst/replaygain/gstrgvolume.c | 698 | ||||
| -rw-r--r-- | gst/replaygain/gstrgvolume.h | 88 | ||||
| -rw-r--r-- | gst/replaygain/replaygain.c | 53 | ||||
| -rw-r--r-- | gst/replaygain/replaygain.h | 36 | ||||
| -rw-r--r-- | gst/replaygain/rganalysis.c | 777 | ||||
| -rw-r--r-- | gst/replaygain/rganalysis.h | 56 |
11 files changed, 0 insertions, 2772 deletions
diff --git a/gst/replaygain/Makefile.am b/gst/replaygain/Makefile.am deleted file mode 100644 index a0a3ca5a..00000000 --- a/gst/replaygain/Makefile.am +++ /dev/null @@ -1,21 +0,0 @@ -plugin_LTLIBRARIES = libgstreplaygain.la - -libgstreplaygain_la_SOURCES = \ - gstrganalysis.c \ - gstrglimiter.c \ - gstrgvolume.c \ - replaygain.c \ - rganalysis.c -libgstreplaygain_la_CFLAGS = \ - $(GST_CFLAGS) $(GST_BASE_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) -libgstreplaygain_la_LIBADD = \ - $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstpbutils-0.10 $(LIBM) -libgstreplaygain_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) - -# headers we need but don't want installed -noinst_HEADERS = \ - gstrganalysis.h \ - gstrglimiter.h \ - gstrgvolume.h \ - replaygain.h \ - rganalysis.h 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 <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 - * @see_also: #GstRgVolume - * - * This element analyzes raw audio sample data in accordance with the proposed - * <ulink url="http://replaygain.org">ReplayGain standard</ulink> for - * calculating the ideal replay gain for music tracks and albums. The element - * is designed as a pass-through filter that never modifies any data. As it - * receives an EOS event, it finalizes the ongoing analysis and generates a tag - * list containing the results. It is sent downstream with a tag event and - * posted on the message bus with a tag message. The EOS event is forwarded as - * normal afterwards. Result tag lists at least contain the tags - * #GST_TAG_TRACK_GAIN, #GST_TAG_TRACK_PEAK and #GST_TAG_REFERENCE_LEVEL. - * - * Because the generated metadata tags become available at the end of streams, - * downstream muxer and encoder elements are normally unable to save them in - * their output since they generally save metadata in the file header. - * Therefore, it is often necessary that applications read the results in a bus - * event handler for the tag message. Obtaining the values this way is always - * needed for <link linkend="GstRgAnalysis--num-tracks">album processing</link> - * since the album gain and peak values need to be associated with all tracks of - * an album, not just the last one. - * - * <refsect2> - * <title>Example launch lines</title> - * |[ - * 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 - * <para> - * The above launch line yields a result gain of +6 dB (instead of the expected - * +0 dB). This is not in error, refer to the #GstRgAnalysis:reference-level - * property documentation for more information. - * </para> - * </refsect2> - * <refsect2> - * <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 <gst/gst.h> -#include <gst/base/gstbasetransform.h> - -#include "gstrganalysis.h" -#include "replaygain.h" - -GST_DEBUG_CATEGORY_STATIC (gst_rg_analysis_debug); -#define GST_CAT_DEFAULT gst_rg_analysis_debug - -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; - - /** - * 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 <link linkend="GstRgAnalysis--num-tracks">album - * processing</link>, the element will skip the number of remaining album - * tracks if a full set of tags is found for the first track. If a subsequent - * track of the album is missing tags, processing cannot start again. If this - * is undesired, the application has to scan all files beforehand and enable - * forcing of processing if needed. - */ - g_object_class_install_property (gobject_class, PROP_FORCED, - g_param_spec_boolean ("forced", "Forced", - "Analyze even if ReplayGain tags exist", - FORCED_DEFAULT, G_PARAM_READWRITE)); - /** - * GstRgAnalysis:reference-level: - * - * Reference level [dB]. - * - * Analyzing the ReplayGain pink noise reference waveform computes a result of - * +6 dB instead of the expected 0 dB. This is because the default reference - * level is 89 dB. To obtain values as lined out in the original proposal of - * ReplayGain, set this property to 83. - * - * Almost all software uses 89 dB as a reference however, and this value has - * become the new official value. That is to say, while the change has been - * acclaimed by the author of the ReplayGain proposal, the <ulink - * url="http://replaygain.org">webpage</ulink> is still outdated at the time - * of this writing. - * - * The value was changed because the original proposal recommends a default - * pre-amp value of +6 dB for playback. This seemed a bit odd, as it means - * that the algorithm has the general tendency to produce adjustment values - * that are 6 dB too low. Bumping the reference level by 6 dB compensated for - * this. - * - * The problem of the reference level being ambiguous for lack of concise - * standardization is to be solved by adopting the #GST_TAG_REFERENCE_LEVEL - * tag, which allows to store the used value alongside the gain values. - */ - g_object_class_install_property (gobject_class, PROP_REFERENCE_LEVEL, - g_param_spec_double ("reference-level", "Reference level", - "Reference level [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; -} diff --git a/gst/replaygain/gstrganalysis.h b/gst/replaygain/gstrganalysis.h deleted file mode 100644 index fbf46830..00000000 --- a/gst/replaygain/gstrganalysis.h +++ /dev/null @@ -1,85 +0,0 @@ -/* 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; -}; - -GType gst_rg_analysis_get_type (void); - -G_END_DECLS - -#endif /* __GST_RG_ANALYSIS_H__ */ diff --git a/gst/replaygain/gstrglimiter.c b/gst/replaygain/gstrglimiter.c deleted file mode 100644 index 43c7b01a..00000000 --- a/gst/replaygain/gstrglimiter.c +++ /dev/null @@ -1,202 +0,0 @@ -/* GStreamer ReplayGain limiter - * - * Copyright (C) 2007 Rene Stadler <mail@renestadler.de> - * - * gstrglimiter.c: Element to apply signal compression to raw audio data - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -/** - * SECTION:element-rglimiter - * @see_also: #GstRgVolume - * - * This element applies signal compression/limiting to raw audio data. It - * performs strict hard limiting with soft-knee characteristics, using a - * threshold of -6 dB. This type of filter is mentioned in the proposed <ulink - * url="http://replaygain.org">ReplayGain standard</ulink>. - * - * <refsect2> - * <title>Example launch line</title> - * |[ - * gst-launch filesrc location=filename.ext ! decodebin ! audioconvert \ - * ! rgvolume pre-amp=6.0 headroom=10.0 ! rglimiter \ - * ! audioconvert ! audioresample ! alsasink - * ]|Playback of a file - * </refsect2> - */ - -#ifdef HAVE_CONFIG_H -#include <config.h> -#endif - -#include <gst/gst.h> -#include <math.h> - -#include "gstrglimiter.h" - -GST_DEBUG_CATEGORY_STATIC (gst_rg_limiter_debug); -#define GST_CAT_DEFAULT gst_rg_limiter_debug - -enum -{ - PROP_0, - PROP_ENABLED, -}; - -static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "width = (int) 32, channels = (int) [1, MAX], " - "rate = (int) [1, MAX], endianness = (int) BYTE_ORDER")); - -static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "width = (int) 32, channels = (int) [1, MAX], " - "rate = (int) [1, MAX], endianness = (int) BYTE_ORDER")); - -GST_BOILERPLATE (GstRgLimiter, gst_rg_limiter, GstBaseTransform, - GST_TYPE_BASE_TRANSFORM); - -static void gst_rg_limiter_class_init (GstRgLimiterClass * klass); -static void gst_rg_limiter_init (GstRgLimiter * filter, - GstRgLimiterClass * gclass); - -static void gst_rg_limiter_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_rg_limiter_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static GstFlowReturn gst_rg_limiter_transform_ip (GstBaseTransform * base, - GstBuffer * buf); - -static const GstElementDetails element_details = { - "ReplayGain limiter", - "Filter/Effect/Audio", - "Apply signal compression to raw audio data", - "Ren\xc3\xa9 Stadler <mail@renestadler.de>" -}; - -static void -gst_rg_limiter_base_init (gpointer g_class) -{ - GstElementClass *element_class = g_class; - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_factory)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_factory)); - gst_element_class_set_details (element_class, &element_details); - - GST_DEBUG_CATEGORY_INIT (gst_rg_limiter_debug, "rglimiter", 0, - "ReplayGain limiter element"); -} - -static void -gst_rg_limiter_class_init (GstRgLimiterClass * klass) -{ - GObjectClass *gobject_class; - GstBaseTransformClass *trans_class; - - gobject_class = (GObjectClass *) klass; - - gobject_class->set_property = gst_rg_limiter_set_property; - gobject_class->get_property = gst_rg_limiter_get_property; - - g_object_class_install_property (gobject_class, PROP_ENABLED, - g_param_spec_boolean ("enabled", "Enabled", "Enable processing", TRUE, - G_PARAM_READWRITE)); - - trans_class = GST_BASE_TRANSFORM_CLASS (klass); - trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_limiter_transform_ip); - trans_class->passthrough_on_same_caps = FALSE; -} - -static void -gst_rg_limiter_init (GstRgLimiter * filter, GstRgLimiterClass * gclass) -{ - GstBaseTransform *base = GST_BASE_TRANSFORM (filter); - - gst_base_transform_set_passthrough (base, FALSE); - gst_base_transform_set_gap_aware (base, TRUE); - - filter->enabled = TRUE; -} - -static void -gst_rg_limiter_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstRgLimiter *filter = GST_RG_LIMITER (object); - - switch (prop_id) { - case PROP_ENABLED: - filter->enabled = g_value_get_boolean (value); - gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter), - !filter->enabled); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_rg_limiter_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstRgLimiter *filter = GST_RG_LIMITER (object); - - switch (prop_id) { - case PROP_ENABLED: - g_value_set_boolean (value, filter->enabled); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -#define LIMIT 1.0 -#define THRES 0.5 /* ca. -6 dB */ -#define COMPL 0.5 /* LIMIT - THRESH */ - -static GstFlowReturn -gst_rg_limiter_transform_ip (GstBaseTransform * base, GstBuffer * buf) -{ - GstRgLimiter *filter = GST_RG_LIMITER (base); - gfloat *input; - guint count; - guint i; - - if (!filter->enabled) - return GST_FLOW_OK; - - if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)) - return GST_FLOW_OK; - - input = (gfloat *) GST_BUFFER_DATA (buf); - count = GST_BUFFER_SIZE (buf) / sizeof (gfloat); - - for (i = count; i--;) { - if (*input > THRES) - *input = tanhf ((*input - THRES) / COMPL) * COMPL + THRES; - else if (*input < -THRES) - *input = tanhf ((*input + THRES) / COMPL) * COMPL - THRES; - input++; - } - - return GST_FLOW_OK; -} diff --git a/gst/replaygain/gstrglimiter.h b/gst/replaygain/gstrglimiter.h deleted file mode 100644 index 63bd8049..00000000 --- a/gst/replaygain/gstrglimiter.h +++ /dev/null @@ -1,64 +0,0 @@ -/* GStreamer ReplayGain limiter - * - * Copyright (C) 2007 Rene Stadler <mail@renestadler.de> - * - * gstrglimiter.h: Element to apply signal compression to raw audio data - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifndef __GST_RG_LIMITER_H__ -#define __GST_RG_LIMITER_H__ - -#include <gst/gst.h> -#include <gst/base/gstbasetransform.h> - -#define GST_TYPE_RG_LIMITER \ - (gst_rg_limiter_get_type()) -#define GST_RG_LIMITER(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_LIMITER,GstRgLimiter)) -#define GST_RG_LIMITER_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_LIMITER,GstRgLimiterClass)) -#define GST_IS_RG_LIMITER(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_LIMITER)) -#define GST_IS_RG_LIMITER_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_LIMITER)) - -typedef struct _GstRgLimiter GstRgLimiter; -typedef struct _GstRgLimiterClass GstRgLimiterClass; - -/** - * GstRgLimiter: - * - * Opaque data structure. - */ -struct _GstRgLimiter -{ - GstBaseTransform element; - - /*< private >*/ - - gboolean enabled; -}; - -struct _GstRgLimiterClass -{ - GstBaseTransformClass parent_class; -}; - -GType gst_rg_limiter_get_type (void); - -#endif /* __GST_RG_LIMITER_H__ */ diff --git a/gst/replaygain/gstrgvolume.c b/gst/replaygain/gstrgvolume.c deleted file mode 100644 index 41fe441d..00000000 --- a/gst/replaygain/gstrgvolume.c +++ /dev/null @@ -1,698 +0,0 @@ -/* GStreamer ReplayGain volume adjustment - * - * Copyright (C) 2007 Rene Stadler <mail@renestadler.de> - * - * gstrgvolume.c: Element to apply ReplayGain volume adjustment - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -/** - * SECTION:element-rgvolume - * @see_also: #GstRgLimiter, #GstRgAnalysis - * - * This element applies volume changes to streams as lined out in the proposed - * <ulink url="http://replaygain.org">ReplayGain standard</ulink>. It - * interprets the ReplayGain meta data tags and carries out the adjustment (by - * using a volume element internally). The relevant tags are: - * <itemizedlist> - * <listitem>#GST_TAG_TRACK_GAIN</listitem> - * <listitem>#GST_TAG_TRACK_PEAK</listitem> - * <listitem>#GST_TAG_ALBUM_GAIN</listitem> - * <listitem>#GST_TAG_ALBUM_PEAK</listitem> - * <listitem>#GST_TAG_REFERENCE_LEVEL</listitem> - * </itemizedlist> - * The information carried by these tags must have been calculated beforehand by - * performing the ReplayGain analysis. This is implemented by the <link - * linkend="GstRgAnalysis">rganalysis</link> element. - * - * The signal compression/limiting recommendations outlined in the proposed - * standard are not implemented by this element. This has to be handled by - * separate elements because applications might want to have additional filters - * between the volume adjustment and the limiting stage. A basic limiter is - * included with this plugin: The <link linkend="GstRgLimiter">rglimiter</link> - * element applies -6 dB hard limiting as mentioned in the ReplayGain standard. - * - * <refsect2> - * <title>Example launch line</title> - * |[ - * gst-launch filesrc location=filename.ext ! decodebin ! audioconvert \ - * ! rgvolume ! audioconvert ! audioresample ! alsasink - * ]| Playback of a file - * </refsect2> - */ - -#ifdef HAVE_CONFIG_H -#include <config.h> -#endif - -#include <gst/gst.h> -#include <gst/pbutils/pbutils.h> -#include <math.h> - -#include "gstrgvolume.h" -#include "replaygain.h" - -GST_DEBUG_CATEGORY_STATIC (gst_rg_volume_debug); -#define GST_CAT_DEFAULT gst_rg_volume_debug - -enum -{ - PROP_0, - PROP_ALBUM_MODE, - PROP_HEADROOM, - PROP_PRE_AMP, - PROP_FALLBACK_GAIN, - PROP_TARGET_GAIN, - PROP_RESULT_GAIN -}; - -#define DEFAULT_ALBUM_MODE TRUE -#define DEFAULT_HEADROOM 0.0 -#define DEFAULT_PRE_AMP 0.0 -#define DEFAULT_FALLBACK_GAIN 0.0 - -#define DB_TO_LINEAR(x) pow (10., (x) / 20.) -#define LINEAR_TO_DB(x) (20. * log10 (x)) - -#define GAIN_FORMAT "+.02f dB" -#define PEAK_FORMAT ".06f" - -#define VALID_GAIN(x) ((x) > -60.00 && (x) < 60.00) -#define VALID_PEAK(x) ((x) > 0.) - -/* Same template caps as GstVolume, for I don't like having just ANY caps. */ - -static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, MAX ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) 32; " - "audio/x-raw-int, " - "channels = (int) [ 1, MAX ], " - "rate = (int) [ 1, MAX ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE")); - -static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, MAX ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) 32; " - "audio/x-raw-int, " - "channels = (int) [ 1, MAX ], " - "rate = (int) [ 1, MAX ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE")); - -GST_BOILERPLATE (GstRgVolume, gst_rg_volume, GstBin, GST_TYPE_BIN); - -static void gst_rg_volume_class_init (GstRgVolumeClass * klass); -static void gst_rg_volume_init (GstRgVolume * self, GstRgVolumeClass * gclass); - -static void gst_rg_volume_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_rg_volume_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); -static void gst_rg_volume_dispose (GObject * object); - -static GstStateChangeReturn gst_rg_volume_change_state (GstElement * element, - GstStateChange transition); -static gboolean gst_rg_volume_sink_event (GstPad * pad, GstEvent * event); - -static GstEvent *gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event); -static void gst_rg_volume_reset (GstRgVolume * self); -static void gst_rg_volume_update_gain (GstRgVolume * self); -static inline void gst_rg_volume_determine_gain (GstRgVolume * self, - gdouble * target_gain, gdouble * result_gain); - -static void -gst_rg_volume_base_init (gpointer g_class) -{ - GstElementClass *element_class = g_class; - - static const GstElementDetails element_details = { - "ReplayGain volume", - "Filter/Effect/Audio", - "Apply ReplayGain volume adjustment", - "Ren\xc3\xa9 Stadler <mail@renestadler.de>" - }; - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_template)); - gst_element_class_set_details (element_class, &element_details); - - GST_DEBUG_CATEGORY_INIT (gst_rg_volume_debug, "rgvolume", 0, - "ReplayGain volume element"); -} - -static void -gst_rg_volume_class_init (GstRgVolumeClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *element_class; - GstBinClass *bin_class; - - gobject_class = (GObjectClass *) klass; - - gobject_class->set_property = gst_rg_volume_set_property; - gobject_class->get_property = gst_rg_volume_get_property; - gobject_class->dispose = gst_rg_volume_dispose; - - /** - * GstRgVolume:album-mode: - * - * Whether to prefer album gain over track gain. - * - * If set to %TRUE, use album gain instead of track gain if both are - * available. This keeps the relative loudness levels of tracks from the same - * album intact. - * - * If set to %FALSE, track mode is used instead. This effectively leads to - * more extensive normalization. - * - * If album mode is enabled but the album gain tag is absent in the stream, - * the track gain is used instead. If both gain tags are missing, the value - * of the <link linkend="GstRgVolume--fallback-gain">fallback-gain</link> - * property is used instead. - */ - g_object_class_install_property (gobject_class, PROP_ALBUM_MODE, - g_param_spec_boolean ("album-mode", "Album mode", - "Prefer album over track gain", DEFAULT_ALBUM_MODE, - G_PARAM_READWRITE)); - /** - * GstRgVolume:headroom: - * - * Extra headroom [dB]. This controls the amount by which the output can - * exceed digital full scale. - * - * Only set this to a value greater than 0.0 if signal compression/limiting of - * a suitable form is applied to the output (or output is brought into the - * correct range by some other transformation). - * - * This element internally uses a volume element, which also supports - * operating on integer audio formats. These formats do not allow exceeding - * digital full scale. If extra headroom is used, make sure that the raw - * audio data format is floating point (audio/x-raw-float). Otherwise, - * clipping distortion might be introduced as part of the volume adjustment - * itself. - */ - g_object_class_install_property (gobject_class, PROP_HEADROOM, - g_param_spec_double ("headroom", "Headroom", "Extra headroom [dB]", - 0., 60., DEFAULT_HEADROOM, G_PARAM_READWRITE)); - /** - * GstRgVolume:pre-amp: - * - * Additional gain to apply globally [dB]. This controls the trade-off - * between uniformity of normalization and utilization of available dynamic - * range. - * - * Note that the default value is 0 dB because the ReplayGain reference value - * was adjusted by +6 dB (from 83 to 89 dB). At the time of this writing, the - * <ulink url="http://replaygain.org">webpage</ulink> is still outdated and - * does not reflect this change however. Where the original proposal states - * that a proper default pre-amp value is +6 dB, this translates to the used 0 - * dB. - */ - g_object_class_install_property (gobject_class, PROP_PRE_AMP, - g_param_spec_double ("pre-amp", "Pre-amp", "Extra gain [dB]", - -60., 60., DEFAULT_PRE_AMP, G_PARAM_READWRITE)); - /** - * GstRgVolume:fallback-gain: - * - * Fallback gain [dB] for streams missing ReplayGain tags. - */ - g_object_class_install_property (gobject_class, PROP_FALLBACK_GAIN, - g_param_spec_double ("fallback-gain", "Fallback gain", - "Gain for streams missing tags [dB]", - -60., 60., DEFAULT_FALLBACK_GAIN, G_PARAM_READWRITE)); - /** - * GstRgVolume:result-gain: - * - * Applied gain [dB]. This gain is applied to processed buffer data. - * - * This is set to the <link linkend="GstRgVolume--target-gain">target - * gain</link> if amplification by that amount can be applied safely. - * "Safely" means that the volume adjustment does not inflict clipping - * distortion. Should this not be the case, the result gain is set to an - * appropriately reduced value (by applying peak normalization). The proposed - * standard calls this "clipping prevention". - * - * The difference between target and result gain reflects the necessary amount - * of reduction. Applications can make use of this information to temporarily - * reduce the <link linkend="GstRgVolume--pre-amp">pre-amp</link> for - * subsequent streams, as recommended by the ReplayGain standard. - * - * Note that target and result gain differing for a great majority of streams - * indicates a problem: What happens in this case is that most streams receive - * peak normalization instead of amplification by the ideal replay gain. To - * prevent this, the <link linkend="GstRgVolume--pre-amp">pre-amp</link> has - * to be lowered and/or a limiter has to be used which facilitates the use of - * <link linkend="GstRgVolume--headroom">headroom</link>. - */ - g_object_class_install_property (gobject_class, PROP_RESULT_GAIN, - g_param_spec_double ("result-gain", "Result-gain", "Applied gain [dB]", - -120., 120., 0., G_PARAM_READABLE)); - /** - * GstRgVolume:target-gain: - * - * Applicable gain [dB]. This gain is supposed to be applied. - * - * Depending on the value of the <link - * linkend="GstRgVolume--album-mode">album-mode</link> property and the - * presence of ReplayGain tags in the stream, this is set according to one of - * these simple formulas: - * - * <itemizedlist> - * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + album gain - * of the stream</listitem> - * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + track gain - * of the stream</listitem> - * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + <link - * linkend="GstRgVolume--fallback-gain">fallback gain</link></listitem> - * </itemizedlist> - */ - g_object_class_install_property (gobject_class, PROP_TARGET_GAIN, - g_param_spec_double ("target-gain", "Target-gain", - "Applicable gain [dB]", -120., 120., 0., G_PARAM_READABLE)); - - element_class = (GstElementClass *) klass; - element_class->change_state = GST_DEBUG_FUNCPTR (gst_rg_volume_change_state); - - bin_class = (GstBinClass *) klass; - /* Setting these to NULL makes gst_bin_add and _remove refuse to let anyone - * mess with our internals. */ - bin_class->add_element = NULL; - bin_class->remove_element = NULL; -} - -static void -gst_rg_volume_init (GstRgVolume * self, GstRgVolumeClass * gclass) -{ - GObjectClass *volume_class; - GstPad *volume_pad, *ghost_pad; - - self->album_mode = DEFAULT_ALBUM_MODE; - self->headroom = DEFAULT_HEADROOM; - self->pre_amp = DEFAULT_PRE_AMP; - self->fallback_gain = DEFAULT_FALLBACK_GAIN; - self->target_gain = 0.0; - self->result_gain = 0.0; - - self->volume_element = gst_element_factory_make ("volume", "rgvolume-volume"); - if (G_UNLIKELY (self->volume_element == NULL)) { - GstMessage *msg; - - GST_WARNING_OBJECT (self, "could not create volume element"); - msg = gst_missing_element_message_new (GST_ELEMENT_CAST (self), "volume"); - gst_element_post_message (GST_ELEMENT_CAST (self), msg); - - /* Nothing else to do, we will refuse the state change from NULL to READY to - * indicate that something went very wrong. It is doubtful that someone - * attempts changing our state though, since we end up having no pads! */ - return; - } - - volume_class = G_OBJECT_GET_CLASS (G_OBJECT (self->volume_element)); - self->max_volume = G_PARAM_SPEC_DOUBLE - (g_object_class_find_property (volume_class, "volume"))->maximum; - - GST_BIN_CLASS (parent_class)->add_element (GST_BIN_CAST (self), - self->volume_element); - - volume_pad = gst_element_get_static_pad (self->volume_element, "sink"); - ghost_pad = gst_ghost_pad_new_from_template ("sink", volume_pad, - gst_pad_get_pad_template (volume_pad)); - gst_object_unref (volume_pad); - gst_pad_set_event_function (ghost_pad, gst_rg_volume_sink_event); - gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad); - - volume_pad = gst_element_get_static_pad (self->volume_element, "src"); - ghost_pad = gst_ghost_pad_new_from_template ("src", volume_pad, - gst_pad_get_pad_template (volume_pad)); - gst_object_unref (volume_pad); - gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad); -} - -static void -gst_rg_volume_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstRgVolume *self = GST_RG_VOLUME (object); - - switch (prop_id) { - case PROP_ALBUM_MODE: - self->album_mode = g_value_get_boolean (value); - break; - case PROP_HEADROOM: - self->headroom = g_value_get_double (value); - break; - case PROP_PRE_AMP: - self->pre_amp = g_value_get_double (value); - break; - case PROP_FALLBACK_GAIN: - self->fallback_gain = g_value_get_double (value); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } - - gst_rg_volume_update_gain (self); -} - -static void -gst_rg_volume_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstRgVolume *self = GST_RG_VOLUME (object); - - switch (prop_id) { - case PROP_ALBUM_MODE: - g_value_set_boolean (value, self->album_mode); - break; - case PROP_HEADROOM: - g_value_set_double (value, self->headroom); - break; - case PROP_PRE_AMP: - g_value_set_double (value, self->pre_amp); - break; - case PROP_FALLBACK_GAIN: - g_value_set_double (value, self->fallback_gain); - break; - case PROP_TARGET_GAIN: - g_value_set_double (value, self->target_gain); - break; - case PROP_RESULT_GAIN: - g_value_set_double (value, self->result_gain); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_rg_volume_dispose (GObject * object) -{ - GstRgVolume *self = GST_RG_VOLUME (object); - - if (self->volume_element != NULL) { - /* Manually remove our child using the bin implementation of remove_element. - * This is needed because we prevent gst_bin_remove from working, which the - * parent dispose handler would use if we had any children left. */ - GST_BIN_CLASS (parent_class)->remove_element (GST_BIN_CAST (self), - self->volume_element); - self->volume_element = NULL; - } - - G_OBJECT_CLASS (parent_class)->dispose (object); -} - -static GstStateChangeReturn -gst_rg_volume_change_state (GstElement * element, GstStateChange transition) -{ - GstRgVolume *self = GST_RG_VOLUME (element); - GstStateChangeReturn res; - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - - if (G_UNLIKELY (self->volume_element == NULL)) { - /* Creating our child volume element in _init failed. */ - return GST_STATE_CHANGE_FAILURE; - } - break; - - case GST_STATE_CHANGE_READY_TO_PAUSED: - - gst_rg_volume_reset (self); - break; - - default: - break; - } - - res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - return res; -} - -/* Event function for the ghost sink pad. */ -static gboolean -gst_rg_volume_sink_event (GstPad * pad, GstEvent * event) -{ - GstRgVolume *self; - GstPad *volume_sink_pad; - GstEvent *send_event = event; - gboolean res; - - self = GST_RG_VOLUME (gst_pad_get_parent_element (pad)); - volume_sink_pad = gst_ghost_pad_get_target (GST_GHOST_PAD (pad)); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_TAG: - - GST_LOG_OBJECT (self, "received tag event"); - - send_event = gst_rg_volume_tag_event (self, event); - - if (send_event == NULL) - GST_LOG_OBJECT (self, "all tags handled, dropping event"); - - break; - - case GST_EVENT_EOS: - - gst_rg_volume_reset (self); - break; - - default: - break; - } - - if (G_LIKELY (send_event != NULL)) - res = gst_pad_send_event (volume_sink_pad, send_event); - else - res = TRUE; - - gst_object_unref (volume_sink_pad); - gst_object_unref (self); - return res; -} - -static GstEvent * -gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event) -{ - GstTagList *tag_list; - gboolean has_track_gain, has_track_peak, has_album_gain, has_album_peak; - gboolean has_ref_level; - - g_return_val_if_fail (event != NULL, NULL); - g_return_val_if_fail (GST_EVENT_TYPE (event) == GST_EVENT_TAG, event); - - gst_event_parse_tag (event, &tag_list); - - if (gst_tag_list_is_empty (tag_list)) - return event; - - has_track_gain = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN, - &self->track_gain); - has_track_peak = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK, - &self->track_peak); - has_album_gain = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN, - &self->album_gain); - has_album_peak = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK, - &self->album_peak); - has_ref_level = gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL, - &self->reference_level); - - if (!has_track_gain && !has_track_peak && !has_album_gain && !has_album_peak) - return event; - - if (has_ref_level && (has_track_gain || has_album_gain) - && (ABS (self->reference_level - RG_REFERENCE_LEVEL) > 1.e-6)) { - /* Log a message stating the amount of adjustment that is applied below. */ - GST_DEBUG_OBJECT (self, - "compensating for reference level difference by %" GAIN_FORMAT, - RG_REFERENCE_LEVEL - self->reference_level); - } - if (has_track_gain) { - self->track_gain += RG_REFERENCE_LEVEL - self->reference_level; - } - if (has_album_gain) { - self->album_gain += RG_REFERENCE_LEVEL - self->reference_level; - } - - /* Ignore values that are obviously invalid. */ - if (G_UNLIKELY (has_track_gain && !VALID_GAIN (self->track_gain))) { - GST_DEBUG_OBJECT (self, - "ignoring bogus track gain value %" GAIN_FORMAT, self->track_gain); - has_track_gain = FALSE; - } - if (G_UNLIKELY (has_track_peak && !VALID_PEAK (self->track_peak))) { - GST_DEBUG_OBJECT (self, - "ignoring bogus track peak value %" PEAK_FORMAT, self->track_peak); - has_track_peak = FALSE; - } - if (G_UNLIKELY (has_album_gain && !VALID_GAIN (self->album_gain))) { - GST_DEBUG_OBJECT (self, - "ignoring bogus album gain value %" GAIN_FORMAT, self->album_gain); - has_album_gain = FALSE; - } - if (G_UNLIKELY (has_album_peak && !VALID_PEAK (self->album_peak))) { - GST_DEBUG_OBJECT (self, - "ignoring bogus album peak value %" PEAK_FORMAT, self->album_peak); - has_album_peak = FALSE; - } - - self->has_track_gain |= has_track_gain; - self->has_track_peak |= has_track_peak; - self->has_album_gain |= has_album_gain; - self->has_album_peak |= has_album_peak; - - event = (GstEvent *) gst_mini_object_make_writable (GST_MINI_OBJECT (event)); - gst_event_parse_tag (event, &tag_list); - - gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_GAIN); - gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_PEAK); - gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_GAIN); - gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_PEAK); - gst_tag_list_remove_tag (tag_list, GST_TAG_REFERENCE_LEVEL); - - gst_rg_volume_update_gain (self); - - if (gst_tag_list_is_empty (tag_list)) { - gst_event_unref (event); - event = NULL; - } - - return event; -} - -static void -gst_rg_volume_reset (GstRgVolume * self) -{ - self->has_track_gain = FALSE; - self->has_track_peak = FALSE; - self->has_album_gain = FALSE; - self->has_album_peak = FALSE; - - self->reference_level = RG_REFERENCE_LEVEL; - - gst_rg_volume_update_gain (self); -} - -static void -gst_rg_volume_update_gain (GstRgVolume * self) -{ - gdouble target_gain, result_gain, result_volume; - gboolean target_gain_changed, result_gain_changed; - - gst_rg_volume_determine_gain (self, &target_gain, &result_gain); - - result_volume = DB_TO_LINEAR (result_gain); - - /* Ensure that the result volume is within the range that the volume element - * can handle. Currently, the limit is 10. (+20 dB), which should not be - * restrictive. */ - if (G_UNLIKELY (result_volume > self->max_volume)) { - GST_INFO_OBJECT (self, - "cannot handle result gain of %" GAIN_FORMAT " (%0.6f), adjusting", - result_gain, result_volume); - - result_volume = self->max_volume; - result_gain = LINEAR_TO_DB (result_volume); - } - - /* Direct comparison is OK in this case. */ - if (target_gain == result_gain) { - GST_INFO_OBJECT (self, - "result gain is %" GAIN_FORMAT " (%0.6f), matching target", - result_gain, result_volume); - } else { - GST_INFO_OBJECT (self, - "result gain is %" GAIN_FORMAT " (%0.6f), target is %" GAIN_FORMAT, - result_gain, result_volume, target_gain); - } - - target_gain_changed = (self->target_gain != target_gain); - result_gain_changed = (self->result_gain != result_gain); - - self->target_gain = target_gain; - self->result_gain = result_gain; - - g_object_set (self->volume_element, "volume", result_volume, NULL); - - if (target_gain_changed) - g_object_notify ((GObject *) self, "target-gain"); - if (result_gain_changed) - g_object_notify ((GObject *) self, "result-gain"); -} - -static inline void -gst_rg_volume_determine_gain (GstRgVolume * self, gdouble * target_gain, - gdouble * result_gain) -{ - gdouble gain, peak; - - if (!self->has_track_gain && !self->has_album_gain) { - - GST_DEBUG_OBJECT (self, "using fallback gain"); - gain = self->fallback_gain; - peak = 1.0; - - } else if ((self->album_mode && self->has_album_gain) - || (!self->album_mode && !self->has_track_gain)) { - - gain = self->album_gain; - if (G_LIKELY (self->has_album_peak)) { - peak = self->album_peak; - } else { - GST_DEBUG_OBJECT (self, "album peak missing, assuming 1.0"); - peak = 1.0; - } - /* Falling back from track to album gain shouldn't really happen. */ - if (G_UNLIKELY (!self->album_mode)) - GST_INFO_OBJECT (self, "falling back to album gain"); - - } else { - /* !album_mode && !has_album_gain || album_mode && has_track_gain */ - - gain = self->track_gain; - if (G_LIKELY (self->has_track_peak)) { - peak = self->track_peak; - } else { - GST_DEBUG_OBJECT (self, "track peak missing, assuming 1.0"); - peak = 1.0; - } - if (self->album_mode) - GST_INFO_OBJECT (self, "falling back to track gain"); - } - - gain += self->pre_amp; - - *target_gain = gain; - *result_gain = gain; - - if (LINEAR_TO_DB (peak) + gain > self->headroom) { - *result_gain = LINEAR_TO_DB (1. / peak) + self->headroom; - } -} diff --git a/gst/replaygain/gstrgvolume.h b/gst/replaygain/gstrgvolume.h deleted file mode 100644 index a0a5a8ce..00000000 --- a/gst/replaygain/gstrgvolume.h +++ /dev/null @@ -1,88 +0,0 @@ -/* GStreamer ReplayGain volume adjustment - * - * Copyright (C) 2007 Rene Stadler <mail@renestadler.de> - * - * gstrgvolume.h: Element to apply ReplayGain volume adjustment - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifndef __GST_RG_VOLUME_H__ -#define __GST_RG_VOLUME_H__ - -#include <gst/gst.h> - -G_BEGIN_DECLS - -#define GST_TYPE_RG_VOLUME \ - (gst_rg_volume_get_type()) -#define GST_RG_VOLUME(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_VOLUME,GstRgVolume)) -#define GST_RG_VOLUME_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_VOLUME,GstRgVolumeClass)) -#define GST_IS_RG_VOLUME(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_VOLUME)) -#define GST_IS_RG_VOLUME_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_VOLUME)) - -typedef struct _GstRgVolume GstRgVolume; -typedef struct _GstRgVolumeClass GstRgVolumeClass; - -/** - * GstRgVolume: - * - * Opaque data structure. - */ -struct _GstRgVolume -{ - GstBin bin; - - /*< private >*/ - - GstElement *volume_element; - gdouble max_volume; - - gboolean album_mode; - gdouble headroom; - gdouble pre_amp; - gdouble fallback_gain; - - gdouble target_gain; - gdouble result_gain; - - gdouble track_gain; - gdouble track_peak; - gdouble album_gain; - gdouble album_peak; - - gboolean has_track_gain; - gboolean has_track_peak; - gboolean has_album_gain; - gboolean has_album_peak; - - gdouble reference_level; -}; - -struct _GstRgVolumeClass -{ - GstBinClass parent_class; -}; - -GType gst_rg_volume_get_type (void); - -G_END_DECLS - -#endif /* __GST_RG_VOLUME_H__ */ diff --git a/gst/replaygain/replaygain.c b/gst/replaygain/replaygain.c deleted file mode 100644 index d0127e8b..00000000 --- a/gst/replaygain/replaygain.c +++ /dev/null @@ -1,53 +0,0 @@ -/* GStreamer ReplayGain plugin - * - * Copyright (C) 2006 Rene Stadler <mail@renestadler.de> - * - * replaygain.c: Plugin providing ReplayGain related elements - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifdef HAVE_CONFIG_H -#include <config.h> -#endif - -#include <gst/gst.h> - -#include "gstrganalysis.h" -#include "gstrglimiter.h" -#include "gstrgvolume.h" - -static gboolean -plugin_init (GstPlugin * plugin) -{ - if (!gst_element_register (plugin, "rganalysis", GST_RANK_NONE, - GST_TYPE_RG_ANALYSIS)) - return FALSE; - - if (!gst_element_register (plugin, "rglimiter", GST_RANK_NONE, - GST_TYPE_RG_LIMITER)) - return FALSE; - - if (!gst_element_register (plugin, "rgvolume", GST_RANK_NONE, - GST_TYPE_RG_VOLUME)) - return FALSE; - - return TRUE; -} - -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "replaygain", - "ReplayGain volume normalization", plugin_init, VERSION, GST_LICENSE, - GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN); diff --git a/gst/replaygain/replaygain.h b/gst/replaygain/replaygain.h deleted file mode 100644 index 15be8885..00000000 --- a/gst/replaygain/replaygain.h +++ /dev/null @@ -1,36 +0,0 @@ -/* GStreamer ReplayGain plugin - * - * Copyright (C) 2006 Rene Stadler <mail@renestadler.de> - * - * replaygain.h: Plugin providing ReplayGain related elements - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifndef __REPLAYGAIN_H__ -#define __REPLAYGAIN_H__ - -G_BEGIN_DECLS - -/* Reference level (in dBSPL). The 2001 proposal specifies 83. This was - * changed later in all implementations to 89, which is the new, offical value: - * David Robinson acknowledged the change but didn't update the website yet. */ - -#define RG_REFERENCE_LEVEL 89. - -G_END_DECLS - -#endif /* __REPLAYGAIN_H__ */ diff --git a/gst/replaygain/rganalysis.c b/gst/replaygain/rganalysis.c deleted file mode 100644 index 147eef85..00000000 --- a/gst/replaygain/rganalysis.c +++ /dev/null @@ -1,777 +0,0 @@ -/* 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 _MSC_VER -/* Disable double-to-float warning: */ -/* A better solution would be to append 'f' to each constant, but that - * makes the code ugly. */ -#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 _MSC_VER -#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) -{ - /* 1e-10 is added below to avoid running into denormals when operating on - * near silence. */ - - output[0] = 1e-10 + 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); -} diff --git a/gst/replaygain/rganalysis.h b/gst/replaygain/rganalysis.h deleted file mode 100644 index 16247361..00000000 --- a/gst/replaygain/rganalysis.h +++ /dev/null @@ -1,56 +0,0 @@ -/* 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 - -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__ */ |