From edc2785eeb8304b58cd4122d9a228d093f6c1c79 Mon Sep 17 00:00:00 2001 From: =?utf8?q?Ren=C3=A9=20Stadler?= Date: Fri, 6 Oct 2006 15:56:01 +0000 Subject: [PATCH] Add ReplayGain analysis element (#357069). MIME-Version: 1.0 Content-Type: text/plain; charset=utf8 Content-Transfer-Encoding: 8bit Original commit message from CVS: Patch by: René Stadler * configure.ac: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * gst/replaygain/Makefile.am: * gst/replaygain/gstrganalysis.c: (gst_rg_analysis_base_init), (gst_rg_analysis_class_init), (gst_rg_analysis_init), (gst_rg_analysis_set_property), (gst_rg_analysis_get_property), (gst_rg_analysis_start), (gst_rg_analysis_set_caps), (gst_rg_analysis_transform_ip), (gst_rg_analysis_event), (gst_rg_analysis_stop), (gst_rg_analysis_handle_tags), (gst_rg_analysis_handle_eos), (gst_rg_analysis_track_result), (gst_rg_analysis_album_result), (plugin_init): * gst/replaygain/gstrganalysis.h: * gst/replaygain/rganalysis.c: (yule_filter), (butter_filter), (apply_filters), (reset_filters), (accumulator_add), (accumulator_clear), (accumulator_result), (rg_analysis_new), (rg_analysis_set_sample_rate), (rg_analysis_destroy), (rg_analysis_analyze_mono_float), (rg_analysis_analyze_stereo_float), (rg_analysis_analyze_mono_int16), (rg_analysis_analyze_stereo_int16), (rg_analysis_analyze), (rg_analysis_track_result), (rg_analysis_album_result), (rg_analysis_reset_album), (rg_analysis_reset): * gst/replaygain/rganalysis.h: Add ReplayGain analysis element (#357069). * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/rganalysis.c: (get_expected_gain), (setup_rganalysis), (cleanup_rganalysis), (set_playing_state), (send_eos_event), (send_tag_event), (poll_eos), (poll_tags), (fail_unless_track_gain), (fail_unless_track_peak), (fail_unless_album_gain), (fail_unless_album_peak), (fail_if_track_tags), (fail_if_album_tags), (fail_unless_num_tracks), (test_buffer_const_float_mono), (test_buffer_const_float_stereo), (test_buffer_const_int16_mono), (test_buffer_const_int16_stereo), (test_buffer_square_float_mono), (test_buffer_square_float_stereo), (test_buffer_square_int16_mono), (test_buffer_square_int16_stereo), (push_buffer), (GST_START_TEST), (rganalysis_suite), (main): Unit tests for the new replaygain element. --- gst/replaygain/Makefile.am | 13 + gst/replaygain/gstrganalysis.c | 686 ++++++++++++++ gst/replaygain/gstrganalysis.h | 83 ++ gst/replaygain/rganalysis.c | 772 +++++++++++++++ gst/replaygain/rganalysis.h | 58 ++ tests/check/elements/rganalysis.c | 1871 +++++++++++++++++++++++++++++++++++++ 6 files changed, 3483 insertions(+) create mode 100644 gst/replaygain/Makefile.am create mode 100644 gst/replaygain/gstrganalysis.c create mode 100644 gst/replaygain/gstrganalysis.h create mode 100644 gst/replaygain/rganalysis.c create mode 100644 gst/replaygain/rganalysis.h create mode 100644 tests/check/elements/rganalysis.c diff --git a/gst/replaygain/Makefile.am b/gst/replaygain/Makefile.am new file mode 100644 index 0000000..eb44591 --- /dev/null +++ b/gst/replaygain/Makefile.am @@ -0,0 +1,13 @@ +plugin_LTLIBRARIES = libgstreplaygain.la + +libgstreplaygain_la_SOURCES = \ + gstrganalysis.c \ + rganalysis.c +libgstreplaygain_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS) +libgstreplaygain_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) +libgstreplaygain_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) + +# headers we need but don't want installed +noinst_HEADERS = \ + gstrganalysis.h \ + rganalysis.h diff --git a/gst/replaygain/gstrganalysis.c b/gst/replaygain/gstrganalysis.c new file mode 100644 index 0000000..adf8555 --- /dev/null +++ b/gst/replaygain/gstrganalysis.c @@ -0,0 +1,686 @@ +/* GStreamer ReplayGain analysis + * + * Copyright (C) 2006 Rene Stadler + * + * gstrganalysis.c: Element that performs the ReplayGain analysis + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public License + * as published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +/** + * SECTION:element-rganalysis + * + * + * + * GstRgAnalysis analyzes raw audio sample data in accordance with the + * proposed ReplayGain + * standard for calculating the ideal replay gain for music + * tracks and albums. The element is designed as a pass-through + * filter that never modifies any data. As it receives an EOS event, + * it finalizes the ongoing analysis and generates a tag list + * containing the results. It is sent downstream with a TAG event and + * posted on the message bus with a TAG message. The EOS event is + * forwarded as normal afterwards. Result tag lists at least contain + * the tags #GST_TAG_TRACK_GAIN and #GST_TAG_TRACK_PEAK. + * + * Album processing + * + * Analyzing several streams sequentially and assigning them a common + * result gain is known as "album processing". If this gain is used + * during playback (by switching to "album mode"), all tracks receive + * the same amplification. This keeps the relative volume levels + * between the tracks intact. To enable this, set the num-tracks property to + * the number of streams that will be processed as album tracks. + * Every time an EOS event is received, the value of this property + * will be decremented by one. As it reaches zero, it is assumed that + * the last track of the album finished. The tag list for the final + * stream will contain the additional tags #GST_TAG_ALBUM_GAIN and + * #GST_TAG_ALBUM_PEAK. All other streams just get the two track tags + * posted because the values for the album tags are not known before + * all tracks are analyzed. Applications need to make sure that the + * album gain and peak values are also associated with the other + * tracks when storing the results. It is thus a bit more complex to + * implement, but should not be avoided since the album gain is + * generally more valuable for use during playback than the track + * gain. + * + * Skipping processing + * + * For assisting transcoder/converter applications, the element can + * silently skip the processing of streams that already contain the + * necessary meta data tags. Data will flow as usual but the element + * will not consume CPU time and will not generate result tags. To + * enable possible skipping, set the forced property to #FALSE. + * If used in conjunction with album processing, the element will skip + * the number of remaining album tracks if a full set of tags is found + * for the first track. If a subsequent track of the album is missing + * tags, processing cannot start again. If this is undesired, your + * application has to scan all files beforehand and enable forcing of + * processing if needed. + * + * Tips + * + * + * Because the generated metadata tags become available at the end of + * streams, downstream muxer and encoder elements are normally unable + * to save them in their output since they generally save metadata in + * the file header. Therefore, it is often necessary that + * applications read the results in a bus event handler for the tag + * message. Obtaining the values this way is always needed for album + * processing since the album gain and peak values need to be + * associated with all tracks of an album, not just the last one. + * + * + * To perform album processing, the element has to preserve data + * between streams. This cannot survive a state change to the NULL or + * READY state. If you change your pipeline's state to NULL or READY + * between tracks, lock the rganalysis element's state using + * gst_element_set_locked_state() when it is in PAUSED or PLAYING. As + * with any other element, don't forget to unlock it again and set it + * to the NULL state before dropping the last reference. + * + * + * If the total number of album tracks is unknown beforehand, set the + * num-tracks property to some large value like #G_MAXINT (or set it + * to >= 2 before each track starts). Before the last track ends, set + * the property value to 1. + * + * + * Compliance + * + * Analyzing the ReplayGain pink noise reference waveform will compute + * a result of +6.00dB instead of the expected 0.00dB because the + * default reference level is 89dB. To obtain values as lined out in + * the original proposal of ReplayGain, set the reference-level + * property to 83. Almost all software uses 89dB as a reference + * however, which works against the tendency of the algorithm to + * advise to drastically lower the volume of music with a highly + * compressed dynamic range and high average output levels. This + * tendency is normally to be fought during playback (if wanted), by + * using a default pre-amp value of at least +6.00dB. At one point, + * the majority of analyzer implementations switched to 89dB which + * moved this adjustment to the analyzing/metadata writing process. + * This change has been acknowledged by the author of the ReplayGain + * proposal, however at the time of this writing, the webpage is still + * not updated. + * + * Example launch lines + * Analyze a simple test waveform: + * + * gst-launch -t audiotestsrc wave=sine num-buffers=512 ! rganalysis ! fakesink + * + * Analyze a given file: + * + * gst-launch -t filesrc location="Some file.ogg" ! decodebin ! audioconvert ! audioresample ! rganalysis ! fakesink + * + * Analyze the pink noise reference file: + * + * gst-launch -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav ! wavparse ! rganalysis ! fakesink + * + * Acknowledgements + * + * This element is based on code used in the vorbisgain program + * and many others. The relevant parts are copyrighted by David + * Robinson, Glen Sawyer and Frank Klemm. + * + * + */ + +#ifdef HAVE_CONFIG_H +#include +#endif + +#include +#include +#include + +#include "gstrganalysis.h" + +GST_DEBUG_CATEGORY_STATIC (gst_rg_analysis_debug); +#define GST_CAT_DEFAULT gst_rg_analysis_debug + +static const GstElementDetails rganalysis_details = { + "ReplayGain analysis", + "Filter/Analyzer/Audio", + "Perform the ReplayGain analysis", + "Ren\xc3\xa9 Stadler " +}; + +/* Default property value. */ +#define FORCED_DEFAULT TRUE + +enum +{ + PROP_0, + PROP_NUM_TRACKS, + PROP_FORCED, + PROP_REFERENCE_LEVEL +}; + +/* The ReplayGain algorithm is intended for use with mono and stereo + * audio. The used implementation has filter coefficients for the + * "usual" sample rates in the 8000 to 48000 Hz range. */ +#define REPLAY_GAIN_CAPS \ + "channels = (int) { 1, 2 }, " \ + "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, " \ + "44100, 48000 }" + +static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " + "width = (int) 32, " "endianness = (int) BYTE_ORDER, " + REPLAY_GAIN_CAPS "; " + "audio/x-raw-int, " + "width = (int) 16, " "depth = (int) [ 1, 16 ], " + "signed = (boolean) true, " "endianness = (int) BYTE_ORDER, " + REPLAY_GAIN_CAPS)); + +static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " + "width = (int) 32, " "endianness = (int) BYTE_ORDER, " + REPLAY_GAIN_CAPS "; " + "audio/x-raw-int, " + "width = (int) 16, " "depth = (int) [ 1, 16 ], " + "signed = (boolean) true, " "endianness = (int) BYTE_ORDER, " + REPLAY_GAIN_CAPS)); + +GST_BOILERPLATE (GstRgAnalysis, gst_rg_analysis, GstBaseTransform, + GST_TYPE_BASE_TRANSFORM); + +static void gst_rg_analysis_class_init (GstRgAnalysisClass * klass); +static void gst_rg_analysis_init (GstRgAnalysis * filter, + GstRgAnalysisClass * gclass); + +static void gst_rg_analysis_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec); +static void gst_rg_analysis_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec); + +static gboolean gst_rg_analysis_start (GstBaseTransform * base); +static gboolean gst_rg_analysis_set_caps (GstBaseTransform * base, + GstCaps * incaps, GstCaps * outcaps); +static GstFlowReturn gst_rg_analysis_transform_ip (GstBaseTransform * base, + GstBuffer * buf); +static gboolean gst_rg_analysis_event (GstBaseTransform * base, + GstEvent * event); +static gboolean gst_rg_analysis_stop (GstBaseTransform * base); + +static void gst_rg_analysis_handle_tags (GstRgAnalysis * filter, + const GstTagList * tag_list); +static void gst_rg_analysis_handle_eos (GstRgAnalysis * filter); +static gboolean gst_rg_analysis_track_result (GstRgAnalysis * filter, + GstTagList ** tag_list); +static gboolean gst_rg_analysis_album_result (GstRgAnalysis * filter, + GstTagList ** tag_list); + +static void +gst_rg_analysis_base_init (gpointer g_class) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); + + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&src_factory)); + gst_element_class_add_pad_template (element_class, + gst_static_pad_template_get (&sink_factory)); + gst_element_class_set_details (element_class, &rganalysis_details); + + GST_DEBUG_CATEGORY_INIT (gst_rg_analysis_debug, "rganalysis", 0, + "ReplayGain analysis element"); +} + +static void +gst_rg_analysis_class_init (GstRgAnalysisClass * klass) +{ + GObjectClass *gobject_class; + GstBaseTransformClass *trans_class; + + gobject_class = (GObjectClass *) klass; + gobject_class->set_property = gst_rg_analysis_set_property; + gobject_class->get_property = gst_rg_analysis_get_property; + + g_object_class_install_property (gobject_class, PROP_NUM_TRACKS, + g_param_spec_int ("num-tracks", "Number of album tracks", + "Number of remaining tracks in the album", + 0, G_MAXINT, 0, G_PARAM_READWRITE)); + g_object_class_install_property (gobject_class, PROP_FORCED, + g_param_spec_boolean ("forced", "Force processing", + "Analyze streams even when ReplayGain tags exist", + FORCED_DEFAULT, G_PARAM_READWRITE)); + g_object_class_install_property (gobject_class, PROP_REFERENCE_LEVEL, + g_param_spec_double ("reference-level", "Reference level", + "Reference level in dB (83.0 for original proposal)", + 0.0, G_MAXDOUBLE, RG_REFERENCE_LEVEL, G_PARAM_READWRITE)); + + trans_class = (GstBaseTransformClass *) klass; + trans_class->start = GST_DEBUG_FUNCPTR (gst_rg_analysis_start); + trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_rg_analysis_set_caps); + trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_analysis_transform_ip); + trans_class->event = GST_DEBUG_FUNCPTR (gst_rg_analysis_event); + trans_class->stop = GST_DEBUG_FUNCPTR (gst_rg_analysis_stop); + trans_class->passthrough_on_same_caps = TRUE; +} + +static void +gst_rg_analysis_init (GstRgAnalysis * filter, GstRgAnalysisClass * gclass) +{ + filter->num_tracks = 0; + filter->forced = FORCED_DEFAULT; + filter->reference_level = RG_REFERENCE_LEVEL; + + filter->ctx = NULL; + filter->analyze = NULL; +} + +static void +gst_rg_analysis_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstRgAnalysis *filter = GST_RG_ANALYSIS (object); + + switch (prop_id) { + case PROP_NUM_TRACKS: + filter->num_tracks = g_value_get_int (value); + break; + case PROP_FORCED: + filter->forced = g_value_get_boolean (value); + break; + case PROP_REFERENCE_LEVEL: + filter->reference_level = g_value_get_double (value); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_rg_analysis_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstRgAnalysis *filter = GST_RG_ANALYSIS (object); + + switch (prop_id) { + case PROP_NUM_TRACKS: + g_value_set_int (value, filter->num_tracks); + break; + case PROP_FORCED: + g_value_set_boolean (value, filter->forced); + break; + case PROP_REFERENCE_LEVEL: + g_value_set_double (value, filter->reference_level); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static gboolean +gst_rg_analysis_start (GstBaseTransform * base) +{ + GstRgAnalysis *filter = GST_RG_ANALYSIS (base); + + filter->ignore_tags = FALSE; + filter->skip = FALSE; + filter->has_track_gain = FALSE; + filter->has_track_peak = FALSE; + filter->has_album_gain = FALSE; + filter->has_album_peak = FALSE; + + filter->ctx = rg_analysis_new (); + filter->analyze = NULL; + + GST_DEBUG_OBJECT (filter, "Started"); + + return TRUE; +} + +static gboolean +gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps, + GstCaps * out_caps) +{ + GstRgAnalysis *filter = GST_RG_ANALYSIS (base); + GstStructure *structure; + const gchar *mime_type; + gint n_channels, sample_rate, sample_bit_size, sample_size; + + g_return_val_if_fail (filter->ctx != NULL, FALSE); + + GST_DEBUG_OBJECT (filter, + "set_caps in %" GST_PTR_FORMAT " out %" GST_PTR_FORMAT, + in_caps, out_caps); + + structure = gst_caps_get_structure (in_caps, 0); + mime_type = gst_structure_get_name (structure); + + if (!gst_structure_get_int (structure, "width", &sample_bit_size) + || !gst_structure_get_int (structure, "channels", &n_channels) + || !gst_structure_get_int (structure, "rate", &sample_rate)) + goto invalid_format; + + if (!rg_analysis_set_sample_rate (filter->ctx, sample_rate)) + goto invalid_format; + + if (sample_bit_size % 8 != 0) + goto invalid_format; + sample_size = sample_bit_size / 8; + + if (strcmp (mime_type, "audio/x-raw-float") == 0) { + + if (sample_size != sizeof (gfloat)) + goto invalid_format; + + /* The depth is not variable for float formats of course. It just + * makes the transform function nice and simple if the + * rg_analysis_analyze_* functions have a common signature. */ + filter->depth = sizeof (gfloat) * 8; + + if (n_channels == 1) + filter->analyze = rg_analysis_analyze_mono_float; + else if (n_channels == 2) + filter->analyze = rg_analysis_analyze_stereo_float; + else + goto invalid_format; + + } else if (strcmp (mime_type, "audio/x-raw-int") == 0) { + + if (sample_size != sizeof (gint16)) + goto invalid_format; + + if (!gst_structure_get_int (structure, "depth", &filter->depth)) + goto invalid_format; + if (filter->depth < 1 || filter->depth > 16) + goto invalid_format; + + if (n_channels == 1) + filter->analyze = rg_analysis_analyze_mono_int16; + else if (n_channels == 2) + filter->analyze = rg_analysis_analyze_stereo_int16; + else + goto invalid_format; + + } else { + + goto invalid_format; + } + + return TRUE; + + /* Errors. */ +invalid_format: + { + filter->analyze = NULL; + GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION, + ("Invalid incoming caps: %" GST_PTR_FORMAT, in_caps), (NULL)); + return FALSE; + } +} + +static GstFlowReturn +gst_rg_analysis_transform_ip (GstBaseTransform * base, GstBuffer * buf) +{ + GstRgAnalysis *filter = GST_RG_ANALYSIS (base); + + g_return_val_if_fail (filter->ctx != NULL, GST_FLOW_ERROR); + g_return_val_if_fail (filter->analyze != NULL, GST_FLOW_ERROR); + + if (filter->skip) + return GST_FLOW_OK; + + GST_DEBUG_OBJECT (filter, "Processing buffer of size %u", + GST_BUFFER_SIZE (buf)); + + filter->analyze (filter->ctx, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), + filter->depth); + + return GST_FLOW_OK; +} + +static gboolean +gst_rg_analysis_event (GstBaseTransform * base, GstEvent * event) +{ + GstRgAnalysis *filter = GST_RG_ANALYSIS (base); + + g_return_val_if_fail (filter->ctx != NULL, TRUE); + + switch (GST_EVENT_TYPE (event)) { + + case GST_EVENT_EOS: + { + GST_DEBUG_OBJECT (filter, "Received EOS event"); + + gst_rg_analysis_handle_eos (filter); + + GST_DEBUG_OBJECT (filter, "Passing on EOS event"); + + break; + } + case GST_EVENT_TAG: + { + GstTagList *tag_list; + + /* The reference to the tag list is borrowed. */ + gst_event_parse_tag (event, &tag_list); + gst_rg_analysis_handle_tags (filter, tag_list); + + break; + } + default: + break; + } + + return TRUE; +} + +static gboolean +gst_rg_analysis_stop (GstBaseTransform * base) +{ + GstRgAnalysis *filter = GST_RG_ANALYSIS (base); + + g_return_val_if_fail (filter->ctx != NULL, FALSE); + + rg_analysis_destroy (filter->ctx); + filter->ctx = NULL; + + GST_DEBUG_OBJECT (filter, "Stopped"); + + return TRUE; +} + +static void +gst_rg_analysis_handle_tags (GstRgAnalysis * filter, + const GstTagList * tag_list) +{ + gboolean album_processing = (filter->num_tracks > 0); + gdouble dummy; + + if (!album_processing) + filter->ignore_tags = FALSE; + + if (filter->skip && album_processing) { + GST_INFO_OBJECT (filter, "Ignoring TAG event: Skipping album"); + return; + } else if (filter->skip) { + GST_INFO_OBJECT (filter, "Ignoring TAG event: Skipping track"); + return; + } else if (filter->ignore_tags) { + GST_INFO_OBJECT (filter, "Ignoring TAG event: Cannot skip anyways"); + return; + } + + filter->has_track_gain |= gst_tag_list_get_double (tag_list, + GST_TAG_TRACK_GAIN, &dummy); + filter->has_track_peak |= gst_tag_list_get_double (tag_list, + GST_TAG_TRACK_PEAK, &dummy); + filter->has_album_gain |= gst_tag_list_get_double (tag_list, + GST_TAG_ALBUM_GAIN, &dummy); + filter->has_album_peak |= gst_tag_list_get_double (tag_list, + GST_TAG_ALBUM_PEAK, &dummy); + + if (!(filter->has_track_gain && filter->has_track_peak)) { + GST_INFO_OBJECT (filter, "Track tags not complete yet"); + return; + } + + if (album_processing && !(filter->has_album_gain && filter->has_album_peak)) { + GST_INFO_OBJECT (filter, "Album tags not complete yet"); + return; + } + + if (filter->forced) { + GST_INFO_OBJECT (filter, + "Existing tags are sufficient, but processing anyway (forced)"); + return; + } + + filter->skip = TRUE; + rg_analysis_reset (filter->ctx); + + if (!album_processing) + GST_INFO_OBJECT (filter, + "Existing tags are sufficient, will not process this track"); + else + GST_INFO_OBJECT (filter, + "Existing tags are sufficient, will not process this album"); +} + +static void +gst_rg_analysis_handle_eos (GstRgAnalysis * filter) +{ + gboolean album_processing = (filter->num_tracks > 0); + gboolean album_finished = (filter->num_tracks == 1); + gboolean album_skipping = album_processing && filter->skip; + + filter->has_track_gain = FALSE; + filter->has_track_peak = FALSE; + + if (album_finished) { + filter->ignore_tags = FALSE; + filter->skip = FALSE; + filter->has_album_gain = FALSE; + filter->has_album_peak = FALSE; + } else if (!album_skipping) { + filter->skip = FALSE; + } + + /* We might have just fully processed a track because it has + * incomplete tags. If we do album processing and allow skipping + * (not forced), prevent switching to skipping if a later track with + * full tags comes along: */ + if (!filter->forced && album_processing && !album_finished) + filter->ignore_tags = TRUE; + + if (!filter->skip) { + GstTagList *tag_list = NULL; + gboolean track_success; + gboolean album_success = FALSE; + + track_success = gst_rg_analysis_track_result (filter, &tag_list); + + if (album_finished) + album_success = gst_rg_analysis_album_result (filter, &tag_list); + else if (!album_processing) + rg_analysis_reset_album (filter->ctx); + + if (track_success || album_success) { + GST_DEBUG_OBJECT (filter, "Posting tag list with results"); + /* This steals our reference to the list: */ + gst_element_found_tags_for_pad (GST_ELEMENT (filter), + GST_BASE_TRANSFORM_SRC_PAD (GST_BASE_TRANSFORM (filter)), tag_list); + } + } + + if (album_processing) { + filter->num_tracks--; + + if (!album_finished) + GST_INFO_OBJECT (filter, "Album not finished yet (num-tracks is now %u)", + filter->num_tracks); + else + GST_INFO_OBJECT (filter, "Album finished (num-tracks is now 0)"); + } + + if (album_processing) + g_object_notify (G_OBJECT (filter), "num-tracks"); +} + +static gboolean +gst_rg_analysis_track_result (GstRgAnalysis * filter, GstTagList ** tag_list) +{ + gboolean track_success; + gdouble track_gain, track_peak; + + track_success = rg_analysis_track_result (filter->ctx, &track_gain, + &track_peak); + + if (track_success) { + track_gain += filter->reference_level - RG_REFERENCE_LEVEL; + GST_INFO_OBJECT (filter, "Track gain is %+.2f dB, peak %.6f", track_gain, + track_peak); + } else { + GST_INFO_OBJECT (filter, "Track was too short to analyze"); + } + + if (track_success) { + if (*tag_list == NULL) + *tag_list = gst_tag_list_new (); + gst_tag_list_add (*tag_list, GST_TAG_MERGE_APPEND, + GST_TAG_TRACK_PEAK, track_peak, GST_TAG_TRACK_GAIN, track_gain, NULL); + } + + return track_success; +} + +static gboolean +gst_rg_analysis_album_result (GstRgAnalysis * filter, GstTagList ** tag_list) +{ + gboolean album_success; + gdouble album_gain, album_peak; + + album_success = rg_analysis_album_result (filter->ctx, &album_gain, + &album_peak); + + if (album_success) { + album_gain += filter->reference_level - RG_REFERENCE_LEVEL; + GST_INFO_OBJECT (filter, "Album gain is %+.2f dB, peak %.6f", album_gain, + album_peak); + } else { + GST_INFO_OBJECT (filter, "Album was too short to analyze"); + } + + if (album_success) { + if (*tag_list == NULL) + *tag_list = gst_tag_list_new (); + gst_tag_list_add (*tag_list, GST_TAG_MERGE_APPEND, + GST_TAG_ALBUM_PEAK, album_peak, GST_TAG_ALBUM_GAIN, album_gain, NULL); + } + + return album_success; +} + +static gboolean +plugin_init (GstPlugin * plugin) +{ + return gst_element_register (plugin, "rganalysis", GST_RANK_NONE, + GST_TYPE_RG_ANALYSIS); +} + +GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "replaygain", + "ReplayGain analysis", plugin_init, VERSION, GST_LICENSE, GST_PACKAGE_NAME, + GST_PACKAGE_ORIGIN); diff --git a/gst/replaygain/gstrganalysis.h b/gst/replaygain/gstrganalysis.h new file mode 100644 index 0000000..121ce4a --- /dev/null +++ b/gst/replaygain/gstrganalysis.h @@ -0,0 +1,83 @@ +/* GStreamer ReplayGain analysis + * + * Copyright (C) 2006 Rene Stadler + * + * 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 +#include + +#include "rganalysis.h" + +G_BEGIN_DECLS + +#define GST_TYPE_RG_ANALYSIS \ + (gst_rg_analysis_get_type()) +#define GST_RG_ANALYSIS(obj) \ + (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_ANALYSIS,GstRgAnalysis)) +#define GST_RG_ANALYSIS_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_ANALYSIS,GstRgAnalysisClass)) +#define GST_IS_RG_ANALYSIS(obj) \ + (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_ANALYSIS)) +#define GST_IS_RG_ANALYSIS_CLASS(klass) \ + (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_ANALYSIS)) +typedef struct _GstRgAnalysis GstRgAnalysis; +typedef struct _GstRgAnalysisClass GstRgAnalysisClass; + +/** + * GstRgAnalysis: + * + * Opaque data structure. + */ +struct _GstRgAnalysis +{ + GstBaseTransform element; + + /*< private >*/ + + RgAnalysisCtx *ctx; + void (*analyze) (RgAnalysisCtx * ctx, gconstpointer data, gsize size, + guint depth); + gint depth; + + /* Property values. */ + guint num_tracks; + gdouble reference_level; + gboolean forced; + + /* State machinery for skipping. */ + gboolean ignore_tags; + gboolean skip; + gboolean has_track_gain; + gboolean has_track_peak; + gboolean has_album_gain; + gboolean has_album_peak; +}; + +struct _GstRgAnalysisClass +{ + GstBaseTransformClass parent_class; +}; + +G_END_DECLS + +#endif /* __GST_RG_ANALYSIS_H__ */ diff --git a/gst/replaygain/rganalysis.c b/gst/replaygain/rganalysis.c new file mode 100644 index 0000000..b20a08f --- /dev/null +++ b/gst/replaygain/rganalysis.c @@ -0,0 +1,772 @@ +/* GStreamer ReplayGain analysis + * + * Copyright (C) 2006 Rene Stadler + * Copyright (C) 2001 David Robinson + * Glen Sawyer + * + * 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 + * and Glen Sawyer , + * 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 +#include +#include + +#include "rganalysis.h" + +#define YULE_ORDER 10 +#define BUTTER_ORDER 2 +/* Percentile which is louder than the proposed level: */ +#define RMS_PERCENTILE 95 +/* Duration of RMS window in milliseconds: */ +#define RMS_WINDOW_MSECS 50 +/* Histogram array elements per dB: */ +#define STEPS_PER_DB 100 +/* Histogram upper bound in dB (normal max. values in the wild are + * assumed to be around 70, 80 dB): */ +#define MAX_DB 120 +/* Calibration value: */ +#define PINK_REF 64.82 /* 298640883795 */ + +#define MAX_ORDER MAX (BUTTER_ORDER, YULE_ORDER) +#define MAX_SAMPLE_RATE 48000 +/* The + 999 has the effect of ceil()ing: */ +#define MAX_SAMPLE_WINDOW (guint) \ + ((MAX_SAMPLE_RATE * RMS_WINDOW_MSECS + 999) / 1000) + +/* Analysis result accumulator. */ + +struct _RgAnalysisAcc +{ + guint32 histogram[STEPS_PER_DB * MAX_DB]; + gdouble peak; +}; + +typedef struct _RgAnalysisAcc RgAnalysisAcc; + +/* Analysis context. */ + +struct _RgAnalysisCtx +{ + /* Filter buffers for left channel. */ + gfloat inprebuf_l[MAX_ORDER * 2]; + gfloat *inpre_l; + gfloat stepbuf_l[MAX_SAMPLE_WINDOW + MAX_ORDER]; + gfloat *step_l; + gfloat outbuf_l[MAX_SAMPLE_WINDOW + MAX_ORDER]; + gfloat *out_l; + /* Filter buffers for right channel. */ + gfloat inprebuf_r[MAX_ORDER * 2]; + gfloat *inpre_r; + gfloat stepbuf_r[MAX_SAMPLE_WINDOW + MAX_ORDER]; + gfloat *step_r; + gfloat outbuf_r[MAX_SAMPLE_WINDOW + MAX_ORDER]; + gfloat *out_r; + + /* Number of samples to reach duration of the RMS window: */ + guint window_n_samples; + /* Progress of the running window: */ + guint window_n_samples_done; + gdouble window_square_sum; + + gint sample_rate; + gint sample_rate_index; + + RgAnalysisAcc track; + RgAnalysisAcc album; +}; + +/* Filter coefficients for the IIR filters that form the equal + * loudness filter. XFilter[ctx->sample_rate_index] gives the array + * of the X coefficients (A or B) for the configured sample rate. */ + +#ifdef G_OS_WIN32 +/* Disable double-to-float warning: */ +#pragma warning ( disable : 4305 ) +#endif + +static const gfloat AYule[9][11] = { + {1., -3.84664617118067, 7.81501653005538, -11.34170355132042, + 13.05504219327545, -12.28759895145294, 9.48293806319790, + -5.87257861775999, 2.75465861874613, -0.86984376593551, + 0.13919314567432}, + {1., -3.47845948550071, 6.36317777566148, -8.54751527471874, 9.47693607801280, + -8.81498681370155, 6.85401540936998, -4.39470996079559, + 2.19611684890774, -0.75104302451432, 0.13149317958808}, + {1., -2.37898834973084, 2.84868151156327, -2.64577170229825, 2.23697657451713, + -1.67148153367602, 1.00595954808547, -0.45953458054983, + 0.16378164858596, -0.05032077717131, 0.02347897407020}, + {1., -1.61273165137247, 1.07977492259970, -0.25656257754070, + -0.16276719120440, -0.22638893773906, 0.39120800788284, + -0.22138138954925, 0.04500235387352, 0.02005851806501, + 0.00302439095741}, + {1., -1.49858979367799, 0.87350271418188, 0.12205022308084, -0.80774944671438, + 0.47854794562326, -0.12453458140019, -0.04067510197014, + 0.08333755284107, -0.04237348025746, 0.02977207319925}, + {1., -0.62820619233671, 0.29661783706366, -0.37256372942400, 0.00213767857124, + -0.42029820170918, 0.22199650564824, 0.00613424350682, 0.06747620744683, + 0.05784820375801, 0.03222754072173}, + {1., -1.04800335126349, 0.29156311971249, -0.26806001042947, 0.00819999645858, + 0.45054734505008, -0.33032403314006, 0.06739368333110, + -0.04784254229033, 0.01639907836189, 0.01807364323573}, + {1., -0.51035327095184, -0.31863563325245, -0.20256413484477, + 0.14728154134330, 0.38952639978999, -0.23313271880868, + -0.05246019024463, -0.02505961724053, 0.02442357316099, + 0.01818801111503}, + {1., -0.25049871956020, -0.43193942311114, -0.03424681017675, + -0.04678328784242, 0.26408300200955, 0.15113130533216, + -0.17556493366449, -0.18823009262115, 0.05477720428674, + 0.04704409688120} +}; + +static const gfloat BYule[9][11] = { + {0.03857599435200, -0.02160367184185, -0.00123395316851, -0.00009291677959, + -0.01655260341619, 0.02161526843274, -0.02074045215285, + 0.00594298065125, 0.00306428023191, 0.00012025322027, 0.00288463683916}, + {0.05418656406430, -0.02911007808948, -0.00848709379851, -0.00851165645469, + -0.00834990904936, 0.02245293253339, -0.02596338512915, + 0.01624864962975, -0.00240879051584, 0.00674613682247, + -0.00187763777362}, + {0.15457299681924, -0.09331049056315, -0.06247880153653, 0.02163541888798, + -0.05588393329856, 0.04781476674921, 0.00222312597743, 0.03174092540049, + -0.01390589421898, 0.00651420667831, -0.00881362733839}, + {0.30296907319327, -0.22613988682123, -0.08587323730772, 0.03282930172664, + -0.00915702933434, -0.02364141202522, -0.00584456039913, + 0.06276101321749, -0.00000828086748, 0.00205861885564, + -0.02950134983287}, + {0.33642304856132, -0.25572241425570, -0.11828570177555, 0.11921148675203, + -0.07834489609479, -0.00469977914380, -0.00589500224440, + 0.05724228140351, 0.00832043980773, -0.01635381384540, + -0.01760176568150}, + {0.44915256608450, -0.14351757464547, -0.22784394429749, -0.01419140100551, + 0.04078262797139, -0.12398163381748, 0.04097565135648, 0.10478503600251, + -0.01863887810927, -0.03193428438915, 0.00541907748707}, + {0.56619470757641, -0.75464456939302, 0.16242137742230, 0.16744243493672, + -0.18901604199609, 0.30931782841830, -0.27562961986224, + 0.00647310677246, 0.08647503780351, -0.03788984554840, + -0.00588215443421}, + {0.58100494960553, -0.53174909058578, -0.14289799034253, 0.17520704835522, + 0.02377945217615, 0.15558449135573, -0.25344790059353, 0.01628462406333, + 0.06920467763959, -0.03721611395801, -0.00749618797172}, + {0.53648789255105, -0.42163034350696, -0.00275953611929, 0.04267842219415, + -0.10214864179676, 0.14590772289388, -0.02459864859345, + -0.11202315195388, -0.04060034127000, 0.04788665548180, + -0.02217936801134} +}; + +static const gfloat AButter[9][3] = { + {1., -1.97223372919527, 0.97261396931306}, + {1., -1.96977855582618, 0.97022847566350}, + {1., -1.95835380975398, 0.95920349965459}, + {1., -1.95002759149878, 0.95124613669835}, + {1., -1.94561023566527, 0.94705070426118}, + {1., -1.92783286977036, 0.93034775234268}, + {1., -1.91858953033784, 0.92177618768381}, + {1., -1.91542108074780, 0.91885558323625}, + {1., -1.88903307939452, 0.89487434461664} +}; + +static const gfloat BButter[9][3] = { + {0.98621192462708, -1.97242384925416, 0.98621192462708}, + {0.98500175787242, -1.97000351574484, 0.98500175787242}, + {0.97938932735214, -1.95877865470428, 0.97938932735214}, + {0.97531843204928, -1.95063686409857, 0.97531843204928}, + {0.97316523498161, -1.94633046996323, 0.97316523498161}, + {0.96454515552826, -1.92909031105652, 0.96454515552826}, + {0.96009142950541, -1.92018285901082, 0.96009142950541}, + {0.95856916599601, -1.91713833199203, 0.95856916599601}, + {0.94597685600279, -1.89195371200558, 0.94597685600279} +}; + +#ifdef G_OS_WIN32 +#pragma warning ( default : 4305 ) +#endif + +/* Filter functions. These access elements with negative indices of + * the input and output arrays (up to the filter's order). */ + +/* For much better performance, the function below has been + * implemented by unrolling the inner loop for our two use cases. */ + +/* + * static inline void + * apply_filter (const gfloat * input, gfloat * output, guint n_samples, + * const gfloat * a, const gfloat * b, guint order) + * { + * gfloat y; + * gint i, k; + * + * for (i = 0; i < n_samples; i++) { + * y = input[i] * b[0]; + * for (k = 1; k <= order; k++) + * y += input[i - k] * b[k] - output[i - k] * a[k]; + * output[i] = y; + * } + * } + */ + +static inline void +yule_filter (const gfloat * input, gfloat * output, + const gfloat * a, const gfloat * b) +{ + output[0] = input[0] * b[0] + + input[-1] * b[1] - output[-1] * a[1] + + input[-2] * b[2] - output[-2] * a[2] + + input[-3] * b[3] - output[-3] * a[3] + + input[-4] * b[4] - output[-4] * a[4] + + input[-5] * b[5] - output[-5] * a[5] + + input[-6] * b[6] - output[-6] * a[6] + + input[-7] * b[7] - output[-7] * a[7] + + input[-8] * b[8] - output[-8] * a[8] + + input[-9] * b[9] - output[-9] * a[9] + + input[-10] * b[10] - output[-10] * a[10]; +} + +static inline void +butter_filter (const gfloat * input, gfloat * output, + const gfloat * a, const gfloat * b) +{ + output[0] = input[0] * b[0] + + input[-1] * b[1] - output[-1] * a[1] + + input[-2] * b[2] - output[-2] * a[2]; +} + +/* Because butter_filter and yule_filter are inlined, this function is + * a bit blown-up (code-size wise), but not inlining gives a ca. 40% + * performance penalty. */ + +static inline void +apply_filters (const RgAnalysisCtx * ctx, const gfloat * input_l, + const gfloat * input_r, guint n_samples) +{ + const gfloat *ayule = AYule[ctx->sample_rate_index]; + const gfloat *byule = BYule[ctx->sample_rate_index]; + const gfloat *abutter = AButter[ctx->sample_rate_index]; + const gfloat *bbutter = BButter[ctx->sample_rate_index]; + gint pos = ctx->window_n_samples_done; + gint i; + + for (i = 0; i < n_samples; i++, pos++) { + yule_filter (input_l + i, ctx->step_l + pos, ayule, byule); + butter_filter (ctx->step_l + pos, ctx->out_l + pos, abutter, bbutter); + + yule_filter (input_r + i, ctx->step_r + pos, ayule, byule); + butter_filter (ctx->step_r + pos, ctx->out_r + pos, abutter, bbutter); + } +} + +/* Clear filter buffer state and current RMS window. */ + +static void +reset_filters (RgAnalysisCtx * ctx) +{ + gint i; + + for (i = 0; i < MAX_ORDER; i++) { + + ctx->inprebuf_l[i] = 0.; + ctx->stepbuf_l[i] = 0.; + ctx->outbuf_l[i] = 0.; + + ctx->inprebuf_r[i] = 0.; + ctx->stepbuf_r[i] = 0.; + ctx->outbuf_r[i] = 0.; + } + + ctx->window_square_sum = 0.; + ctx->window_n_samples_done = 0; +} + +/* Accumulator functions. */ + +/* Add two accumulators in-place. The sum is defined as the result of + * the vector sum of the histogram array and the maximum value of the + * peak field. Thus "adding" the accumulators for all tracks yields + * the correct result for obtaining the album gain and peak. */ + +static void +accumulator_add (RgAnalysisAcc * acc, const RgAnalysisAcc * acc_other) +{ + gint i; + + for (i = 0; i < G_N_ELEMENTS (acc->histogram); i++) + acc->histogram[i] += acc_other->histogram[i]; + + acc->peak = MAX (acc->peak, acc_other->peak); +} + +/* Reset an accumulator to zero. */ + +static void +accumulator_clear (RgAnalysisAcc * acc) +{ + memset (acc->histogram, 0, sizeof (acc->histogram)); + acc->peak = 0.; +} + +/* Obtain final analysis result from an accumulator. Returns TRUE on + * success, FALSE on error (if accumulator is still zero). */ + +static gboolean +accumulator_result (const RgAnalysisAcc * acc, gdouble * result_gain, + gdouble * result_peak) +{ + guint32 sum = 0; + guint32 upper; + guint i; + + for (i = 0; i < G_N_ELEMENTS (acc->histogram); i++) + sum += acc->histogram[i]; + + if (sum == 0) + /* All entries are 0: We got less than 50ms of data. */ + return FALSE; + + upper = (guint32) ceil (sum * (1. - (gdouble) (RMS_PERCENTILE / 100.))); + + for (i = G_N_ELEMENTS (acc->histogram); i--;) { + if (upper <= acc->histogram[i]) + break; + upper -= acc->histogram[i]; + } + + if (result_peak != NULL) + *result_peak = acc->peak; + if (result_gain != NULL) + *result_gain = PINK_REF - (gdouble) i / STEPS_PER_DB; + + return TRUE; +} + +/* Functions that operate on contexts, for external usage. */ + +/* Create a new context. Before it can be used, a sample rate must be + * configured using rg_analysis_set_sample_rate. */ + +RgAnalysisCtx * +rg_analysis_new (void) +{ + RgAnalysisCtx *ctx; + + ctx = g_new (RgAnalysisCtx, 1); + + ctx->inpre_l = ctx->inprebuf_l + MAX_ORDER; + ctx->step_l = ctx->stepbuf_l + MAX_ORDER; + ctx->out_l = ctx->outbuf_l + MAX_ORDER; + + ctx->inpre_r = ctx->inprebuf_r + MAX_ORDER; + ctx->step_r = ctx->stepbuf_r + MAX_ORDER; + ctx->out_r = ctx->outbuf_r + MAX_ORDER; + + ctx->sample_rate = 0; + + accumulator_clear (&ctx->track); + accumulator_clear (&ctx->album); + + return ctx; +} + +/* Adapt to given sample rate. Does nothing if already the current + * rate (returns TRUE then). Returns FALSE only if given sample rate + * is not supported. If the configured rate changes, the last + * unprocessed incomplete 50ms chunk of data is dropped because the + * filters are reset. */ + +gboolean +rg_analysis_set_sample_rate (RgAnalysisCtx * ctx, gint sample_rate) +{ + g_return_val_if_fail (ctx != NULL, FALSE); + + if (ctx->sample_rate == sample_rate) + return TRUE; + + switch (sample_rate) { + case 48000: + ctx->sample_rate_index = 0; + break; + case 44100: + ctx->sample_rate_index = 1; + break; + case 32000: + ctx->sample_rate_index = 2; + break; + case 24000: + ctx->sample_rate_index = 3; + break; + case 22050: + ctx->sample_rate_index = 4; + break; + case 16000: + ctx->sample_rate_index = 5; + break; + case 12000: + ctx->sample_rate_index = 6; + break; + case 11025: + ctx->sample_rate_index = 7; + break; + case 8000: + ctx->sample_rate_index = 8; + break; + default: + return FALSE; + } + + ctx->sample_rate = sample_rate; + /* The + 999 has the effect of ceil()ing: */ + ctx->window_n_samples = (guint) ((sample_rate * RMS_WINDOW_MSECS + 999) + / 1000); + + reset_filters (ctx); + + return TRUE; +} + +void +rg_analysis_destroy (RgAnalysisCtx * ctx) +{ + g_free (ctx); +} + +/* Entry points for analyzing sample data in common raw data formats. + * The stereo format functions expect interleaved frames. It is + * possible to pass data in different formats for the same context, + * there are no restrictions. All functions have the same signature; + * the depth argument for the float functions is not variable and must + * be given the value 32. */ + +void +rg_analysis_analyze_mono_float (RgAnalysisCtx * ctx, gconstpointer data, + gsize size, guint depth) +{ + gfloat conv_samples[512]; + const gfloat *samples = (gfloat *) data; + guint n_samples = size / sizeof (gfloat); + gint i; + + g_return_if_fail (depth == 32); + g_return_if_fail (size % sizeof (gfloat) == 0); + + while (n_samples) { + gint n = MIN (n_samples, G_N_ELEMENTS (conv_samples)); + + n_samples -= n; + memcpy (conv_samples, samples, n * sizeof (gfloat)); + for (i = 0; i < n; i++) { + ctx->track.peak = MAX (ctx->track.peak, fabs (conv_samples[i])); + conv_samples[i] *= 32768.; + } + samples += n; + rg_analysis_analyze (ctx, conv_samples, NULL, n); + } +} + +void +rg_analysis_analyze_stereo_float (RgAnalysisCtx * ctx, gconstpointer data, + gsize size, guint depth) +{ + gfloat conv_samples_l[256]; + gfloat conv_samples_r[256]; + const gfloat *samples = (gfloat *) data; + guint n_frames = size / (sizeof (gfloat) * 2); + gint i; + + g_return_if_fail (depth == 32); + g_return_if_fail (size % (sizeof (gfloat) * 2) == 0); + + while (n_frames) { + gint n = MIN (n_frames, G_N_ELEMENTS (conv_samples_l)); + + n_frames -= n; + for (i = 0; i < n; i++) { + gfloat old_sample; + + old_sample = samples[2 * i]; + ctx->track.peak = MAX (ctx->track.peak, fabs (old_sample)); + conv_samples_l[i] = old_sample * 32768.; + + old_sample = samples[2 * i + 1]; + ctx->track.peak = MAX (ctx->track.peak, fabs (old_sample)); + conv_samples_r[i] = old_sample * 32768.; + } + samples += 2 * n; + rg_analysis_analyze (ctx, conv_samples_l, conv_samples_r, n); + } +} + +void +rg_analysis_analyze_mono_int16 (RgAnalysisCtx * ctx, gconstpointer data, + gsize size, guint depth) +{ + gfloat conv_samples[512]; + gint32 peak_sample = 0; + const gint16 *samples = (gint16 *) data; + guint n_samples = size / sizeof (gint16); + gint shift = sizeof (gint16) * 8 - depth; + gint i; + + g_return_if_fail (depth <= (sizeof (gint16) * 8)); + g_return_if_fail (size % sizeof (gint16) == 0); + + while (n_samples) { + gint n = MIN (n_samples, G_N_ELEMENTS (conv_samples)); + + n_samples -= n; + for (i = 0; i < n; i++) { + gint16 old_sample = samples[i] << shift; + + peak_sample = MAX (peak_sample, ABS ((gint32) old_sample)); + conv_samples[i] = (gfloat) old_sample; + } + samples += n; + rg_analysis_analyze (ctx, conv_samples, NULL, n); + } + ctx->track.peak = MAX (ctx->track.peak, + (gdouble) peak_sample / ((gdouble) (1u << 15))); +} + +void +rg_analysis_analyze_stereo_int16 (RgAnalysisCtx * ctx, gconstpointer data, + gsize size, guint depth) +{ + gfloat conv_samples_l[256]; + gfloat conv_samples_r[256]; + gint32 peak_sample = 0; + const gint16 *samples = (gint16 *) data; + guint n_frames = size / (sizeof (gint16) * 2); + gint shift = sizeof (gint16) * 8 - depth; + gint i; + + g_return_if_fail (depth <= (sizeof (gint16) * 8)); + g_return_if_fail (size % (sizeof (gint16) * 2) == 0); + + while (n_frames) { + gint n = MIN (n_frames, G_N_ELEMENTS (conv_samples_l)); + + n_frames -= n; + for (i = 0; i < n; i++) { + gint16 old_sample; + + old_sample = samples[2 * i] << shift; + peak_sample = MAX (peak_sample, ABS ((gint32) old_sample)); + conv_samples_l[i] = (gfloat) old_sample; + + old_sample = samples[2 * i + 1] << shift; + peak_sample = MAX (peak_sample, ABS ((gint32) old_sample)); + conv_samples_r[i] = (gfloat) old_sample; + } + samples += 2 * n; + rg_analysis_analyze (ctx, conv_samples_l, conv_samples_r, n); + } + ctx->track.peak = MAX (ctx->track.peak, + (gdouble) peak_sample / ((gdouble) (1u << 15))); +} + +/* Analyze the given chunk of samples. The sample data is given in + * floating point format but should be scaled such that the values + * +/-32768.0 correspond to the -0dBFS reference amplitude. + * + * samples_l: Buffer with sample data for the left channel or of the + * mono channel. + * + * samples_r: Buffer with sample data for the right channel or NULL + * for mono. + * + * n_samples: Number of samples passed in each buffer. + */ + +void +rg_analysis_analyze (RgAnalysisCtx * ctx, const gfloat * samples_l, + const gfloat * samples_r, guint n_samples) +{ + const gfloat *input_l, *input_r; + guint n_samples_done; + gint i; + + g_return_if_fail (ctx != NULL); + g_return_if_fail (samples_l != NULL); + g_return_if_fail (ctx->sample_rate != 0); + + if (n_samples == 0) + return; + + if (samples_r == NULL) + /* Mono. */ + samples_r = samples_l; + + memcpy (ctx->inpre_l, samples_l, + MIN (n_samples, MAX_ORDER) * sizeof (gfloat)); + memcpy (ctx->inpre_r, samples_r, + MIN (n_samples, MAX_ORDER) * sizeof (gfloat)); + + n_samples_done = 0; + while (n_samples_done < n_samples) { + /* Limit number of samples to be processed in this iteration to + * the number needed to complete the next window: */ + guint n_samples_current = MIN (n_samples - n_samples_done, + ctx->window_n_samples - ctx->window_n_samples_done); + + if (n_samples_done < MAX_ORDER) { + input_l = ctx->inpre_l + n_samples_done; + input_r = ctx->inpre_r + n_samples_done; + n_samples_current = MIN (n_samples_current, MAX_ORDER - n_samples_done); + } else { + input_l = samples_l + n_samples_done; + input_r = samples_r + n_samples_done; + } + + apply_filters (ctx, input_l, input_r, n_samples_current); + + /* Update the square sum. */ + for (i = 0; i < n_samples_current; i++) + ctx->window_square_sum += ctx->out_l[ctx->window_n_samples_done + i] + * ctx->out_l[ctx->window_n_samples_done + i] + + ctx->out_r[ctx->window_n_samples_done + i] + * ctx->out_r[ctx->window_n_samples_done + i]; + + ctx->window_n_samples_done += n_samples_current; + + g_return_if_fail (ctx->window_n_samples_done <= ctx->window_n_samples); + + if (ctx->window_n_samples_done == ctx->window_n_samples) { + /* Get the Root Mean Square (RMS) for this set of samples. */ + gdouble val = STEPS_PER_DB * 10. * log10 (ctx->window_square_sum / + ctx->window_n_samples * 0.5 + 1.e-37); + gint ival = CLAMP ((gint) val, 0, + (gint) G_N_ELEMENTS (ctx->track.histogram) - 1); + + ctx->track.histogram[ival]++; + ctx->window_square_sum = 0.; + ctx->window_n_samples_done = 0; + + /* No need for memmove here, the areas never overlap: Even for + * the smallest sample rate, the number of samples needed for + * the window is greater than MAX_ORDER. */ + + memcpy (ctx->stepbuf_l, ctx->stepbuf_l + ctx->window_n_samples, + MAX_ORDER * sizeof (gfloat)); + memcpy (ctx->outbuf_l, ctx->outbuf_l + ctx->window_n_samples, + MAX_ORDER * sizeof (gfloat)); + + memcpy (ctx->stepbuf_r, ctx->stepbuf_r + ctx->window_n_samples, + MAX_ORDER * sizeof (gfloat)); + memcpy (ctx->outbuf_r, ctx->outbuf_r + ctx->window_n_samples, + MAX_ORDER * sizeof (gfloat)); + } + + n_samples_done += n_samples_current; + } + + if (n_samples >= MAX_ORDER) { + + memcpy (ctx->inprebuf_l, samples_l + n_samples - MAX_ORDER, + MAX_ORDER * sizeof (gfloat)); + + memcpy (ctx->inprebuf_r, samples_r + n_samples - MAX_ORDER, + MAX_ORDER * sizeof (gfloat)); + + } else { + + memmove (ctx->inprebuf_l, ctx->inprebuf_l + n_samples, + (MAX_ORDER - n_samples) * sizeof (gfloat)); + memcpy (ctx->inprebuf_l + MAX_ORDER - n_samples, samples_l, + n_samples * sizeof (gfloat)); + + memmove (ctx->inprebuf_r, ctx->inprebuf_r + n_samples, + (MAX_ORDER - n_samples) * sizeof (gfloat)); + memcpy (ctx->inprebuf_r + MAX_ORDER - n_samples, samples_r, + n_samples * sizeof (gfloat)); + + } +} + +/* Obtain track gain and peak. Returns TRUE on success. Can fail if + * not enough samples have been processed. Updates album accumulator. + * Resets track accumulator. */ + +gboolean +rg_analysis_track_result (RgAnalysisCtx * ctx, gdouble * gain, gdouble * peak) +{ + gboolean result; + + g_return_val_if_fail (ctx != NULL, FALSE); + + accumulator_add (&ctx->album, &ctx->track); + result = accumulator_result (&ctx->track, gain, peak); + accumulator_clear (&ctx->track); + + reset_filters (ctx); + + return result; +} + +/* Obtain album gain and peak. Returns TRUE on success. Can fail if + * not enough samples have been processed. Resets album + * accumulator. */ + +gboolean +rg_analysis_album_result (RgAnalysisCtx * ctx, gdouble * gain, gdouble * peak) +{ + gboolean result; + + g_return_val_if_fail (ctx != NULL, FALSE); + + result = accumulator_result (&ctx->album, gain, peak); + accumulator_clear (&ctx->album); + + return result; +} + +void +rg_analysis_reset_album (RgAnalysisCtx * ctx) +{ + accumulator_clear (&ctx->album); +} + +/* Reset internal buffers as well as track and album accumulators. + * Configured sample rate is kept intact. */ + +void +rg_analysis_reset (RgAnalysisCtx * ctx) +{ + g_return_if_fail (ctx != NULL); + + reset_filters (ctx); + accumulator_clear (&ctx->track); + accumulator_clear (&ctx->album); +} diff --git a/gst/replaygain/rganalysis.h b/gst/replaygain/rganalysis.h new file mode 100644 index 0000000..39bf9b4 --- /dev/null +++ b/gst/replaygain/rganalysis.h @@ -0,0 +1,58 @@ +/* GStreamer ReplayGain analysis + * + * Copyright (C) 2006 Rene Stadler + * Copyright (C) 2001 David Robinson + * Glen Sawyer + * + * 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 + +G_BEGIN_DECLS + +#define RG_REFERENCE_LEVEL 89. + +typedef struct _RgAnalysisCtx RgAnalysisCtx; + +RgAnalysisCtx *rg_analysis_new (void); +gboolean rg_analysis_set_sample_rate (RgAnalysisCtx * ctx, gint sample_rate); +void rg_analysis_analyze_mono_float (RgAnalysisCtx * ctx, gconstpointer data, + gsize size, guint depth); +void rg_analysis_analyze_stereo_float (RgAnalysisCtx * ctx, gconstpointer data, + gsize size, guint depth); +void rg_analysis_analyze_mono_int16 (RgAnalysisCtx * ctx, gconstpointer data, + gsize size, guint depth); +void rg_analysis_analyze_stereo_int16 (RgAnalysisCtx * ctx, gconstpointer data, + gsize size, guint depth); +void rg_analysis_analyze (RgAnalysisCtx * ctx, const gfloat * samples_l, + const gfloat * samples_r, guint n_samples); +gboolean rg_analysis_track_result (RgAnalysisCtx * ctx, gdouble * gain, + gdouble * peak); +gboolean rg_analysis_album_result (RgAnalysisCtx * ctx, gdouble * gain, + gdouble * peak); +void rg_analysis_reset_album (RgAnalysisCtx * ctx); +void rg_analysis_reset (RgAnalysisCtx * ctx); +void rg_analysis_destroy (RgAnalysisCtx * ctx); + +G_END_DECLS + +#endif /* __RG_ANALYSIS_H__ */ diff --git a/tests/check/elements/rganalysis.c b/tests/check/elements/rganalysis.c new file mode 100644 index 0000000..17b4d62 --- /dev/null +++ b/tests/check/elements/rganalysis.c @@ -0,0 +1,1871 @@ +/* GStreamer ReplayGain analysis + * + * Copyright (C) 2006 Rene Stadler + * + * rganalysis.c: Unit test for the rganalysis element + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Lesser General Public License + * as published by the Free Software Foundation; either version 2.1 of + * the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, but + * WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Lesser General Public License for more details. + * + * You should have received a copy of the GNU Lesser General Public + * License along with this library; if not, write to the Free Software + * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA + * 02110-1301 USA + */ + +/* Some things to note about the RMS window length of the analysis + * algorithm and thus the implementation used in the element: + * Processing divides input data into 50ms windows at some point. + * Some details about this that normally do not matter: + * + * 1. At the end of a stream, the remainder of data that did not fill + * up the last 50ms window is simply discarded. + * + * 2. If the sample rate changes during a stream, the currently + * running window is discarded and the equal loudness filter gets + * reset as if a new stream started. + * + * 3. For the album gain, it is not entirely correct to think of + * obtaining it like "as if all the tracks are analyzed as one + * track". There isn't a separate window being tracked for album + * processing, so at stream (track) end, the remaining unfilled + * window does not contribute to the album gain either. + * + * 4. If a waveform with a result gain G is concatenated to itself + * and the result processed as a track, the gain can be different + * from G if and only if the duration of the original waveform is + * not an integer multiple of 50ms. If the original waveform gets + * processed as a single track and then the same data again as a + * subsequent track, the album result gain will always match G + * (this is implied by 3.). + * + * 5. A stream shorter than 50ms cannot be analyzed. At 8000 and + * 48000 Hz, this corresponds to 400 resp. 2400 frames. If a + * stream is shorter than 50ms, the element will not generate tags + * at EOS (only if an album finished, but only album tags are + * generated then). This is not an erroneous condition, the + * element should behave normally. + * + * The limitations outlined in 1.-4. do not apply to the peak values. + * Every single sample is accounted for when looking for the peak. + * Thus the album peak is guaranteed to be the maximum value of all + * track peaks. + * + * In normal day-to-day use, these little facts are unlikely to be + * relevant, but they have to be kept in mind for writing the tests + * here. + */ + +#include + +GList *buffers = NULL; + +/* For ease of programming we use globals to keep refs for our floating + * src and sink pads we create; otherwise we always have to do get_pad, + * get_peer, and then remove references in every test function */ +static GstPad *mysrcpad, *mysinkpad; + +/* Mapping from supported sample rates to the correct result gain for + * the following test waveform: 20 * 512 samples with a quarter-full + * amplitude of toggling sign, changing every 48 samples and starting + * with the positive value. + * + * Even if we would generate a wave describing a signal with the same + * frequency at each sampling rate, the results would vary (slightly). + * Hence the simple generation method, since we cannot use a constant + * value as expected result anyways. For all sample rates, changing + * the sign every 48 frames gives a sane frequency. Buffers + * containing data that forms such a waveform is created using the + * test_buffer_square_{float,int16}_{mono,stereo} functions below. + * + * The results have been checked against what the metaflac and + * wavegain programs generate for such a stream. If you want to + * verify these, be sure that the metaflac program does not produce + * incorrect results in your environment: I found a strange bug in the + * (defacto) reference code for the analysis that sometimes leads to + * incorrect RMS window lengths. */ + +struct rate_test +{ + guint sample_rate; + gdouble gain; +}; + +static const struct rate_test supported_rates[] = { + 8000, -0.91, + 11025, -2.80, + 12000, -3.13, + 16000, -4.26, + 22050, -5.64, + 24000, -5.87, + 32000, -6.03, + 44100, -6.20, + 48000, -6.14 +}; + +/* Lookup the correct gain adjustment result in above array. */ + +static gdouble +get_expected_gain (guint sample_rate) +{ + gint i; + + for (i = G_N_ELEMENTS (supported_rates); i--;) + if (supported_rates[i].sample_rate == sample_rate) + return supported_rates[i].gain; + g_return_val_if_reached (0.0); +} + +#define SILENCE_GAIN 64.82 + +#define REPLAY_GAIN_CAPS \ + "channels = (int) { 1, 2 }, " \ + "rate = (int) { 8000, 11025, 12000, 16000, 22050, " \ + "24000, 32000, 44100, 48000 }" + +#define RG_ANALYSIS_CAPS_TEMPLATE_STRING \ + "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 sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink", + GST_PAD_SINK, + GST_PAD_ALWAYS, + GST_STATIC_CAPS (RG_ANALYSIS_CAPS_TEMPLATE_STRING) + ); +static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src", + GST_PAD_SRC, + GST_PAD_ALWAYS, + GST_STATIC_CAPS (RG_ANALYSIS_CAPS_TEMPLATE_STRING) + ); + +GstElement * +setup_rganalysis () +{ + GstElement *analysis; + GstBus *bus; + + GST_DEBUG ("setup_rganalysis"); + analysis = gst_check_setup_element ("rganalysis"); + mysrcpad = gst_check_setup_src_pad (analysis, &srctemplate, NULL); + mysinkpad = gst_check_setup_sink_pad (analysis, &sinktemplate, NULL); + gst_pad_set_active (mysrcpad, TRUE); + gst_pad_set_active (mysinkpad, TRUE); + + bus = gst_bus_new (); + gst_element_set_bus (analysis, bus); + /* gst_element_set_bus does not steal a reference. */ + gst_object_unref (bus); + + return analysis; +} + +void +cleanup_rganalysis (GstElement * element) +{ + GST_DEBUG ("cleanup_rganalysis"); + + g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL); + g_list_free (buffers); + buffers = NULL; + + /* The bus owns references to the element: */ + gst_element_set_bus (element, NULL); + + gst_check_teardown_src_pad (element); + gst_check_teardown_sink_pad (element); + gst_check_teardown_element (element); +} + +static void +set_playing_state (GstElement * element) +{ + fail_unless (gst_element_set_state (element, + GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS, + "Could not set state to PLAYING"); +} + +static void +send_eos_event (GstElement * element) +{ + GstBus *bus = gst_element_get_bus (element); + GstPad *pad = gst_element_get_pad (element, "sink"); + GstEvent *event = gst_event_new_eos (); + + fail_unless (gst_pad_send_event (pad, event), + "Cannot send EOS event: Not handled."); + + /* There is no sink element, so _we_ post the EOS message on the bus + * here. Of course we generate any EOS ourselves, but this allows + * us to poll for the EOS message in poll_eos if we expect the + * element to _not_ generate a TAG message. That's better than + * waiting for a timeout to lapse. */ + fail_unless (gst_bus_post (bus, gst_message_new_eos (NULL))); + + gst_object_unref (bus); + gst_object_unref (pad); +} + +static void +send_tag_event (GstElement * element, GstTagList * tag_list) +{ + GstPad *pad = gst_element_get_pad (element, "sink"); + GstEvent *event = gst_event_new_tag (tag_list); + + fail_unless (gst_pad_send_event (pad, event), + "Cannot send TAG event: Not handled."); + + gst_object_unref (pad); +} + +static void +poll_eos (GstElement * element) +{ + GstBus *bus = gst_element_get_bus (element); + GstMessage *message; + + message = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_TAG, GST_SECOND); + fail_unless (message != NULL, "Could not poll for EOS message: Timed out"); + fail_unless (message->type == GST_MESSAGE_EOS, + "Could not poll for eos message: got message of type %s instead", + gst_message_type_get_name (message->type)); + + gst_message_unref (message); + gst_object_unref (bus); +} + +/* This also polls for EOS since the TAG message comes right before + * the end of streams. */ + +static GstTagList * +poll_tags (GstElement * element) +{ + GstBus *bus = gst_element_get_bus (element); + GstTagList *tag_list; + GstMessage *message; + + message = gst_bus_poll (bus, GST_MESSAGE_TAG, GST_SECOND); + fail_unless (message != NULL, "Could not poll for TAG message: Timed out"); + + fail_unless (GST_MESSAGE_SRC (message) == GST_OBJECT (element)); + + gst_message_parse_tag (message, &tag_list); + gst_message_unref (message); + gst_object_unref (bus); + + poll_eos (element); + + return tag_list; +} + +#define MATCH_PEAK(p1, p2) ((p1 < p2 + 1e-6) && (p2 < p1 + 1e-6)) +#define MATCH_GAIN(g1, g2) ((g1 < g2 + 1e-13) && (g2 < g1 + 1e-13)) + +static void +fail_unless_track_gain (const GstTagList * tag_list, gdouble gain) +{ + gdouble result; + + fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN, &result), + "Tag list contains no track gain value"); + fail_unless (MATCH_GAIN (gain, result), + "Track gain %+.2f does not match, expected %+.2f", result, gain); +} + +static void +fail_unless_track_peak (const GstTagList * tag_list, gdouble peak) +{ + gdouble result; + + fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK, &result), + "Tag list contains no track peak value"); + fail_unless (MATCH_PEAK (peak, result), + "Track peak %f does not match, expected %f", result, peak); +} + +static void +fail_unless_album_gain (const GstTagList * tag_list, gdouble gain) +{ + gdouble result; + + fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN, &result), + "Tag list contains no album gain value"); + fail_unless (MATCH_GAIN (result, gain), + "Album gain %+.2f does not match, expected %+.2f", result, gain); +} + +static void +fail_unless_album_peak (const GstTagList * tag_list, gdouble peak) +{ + gdouble result; + + fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK, &result), + "Tag list contains no album peak value"); + fail_unless (MATCH_PEAK (peak, result), + "Album peak %f does not match, expected %f", result, peak); +} + +static void +fail_if_track_tags (const GstTagList * tag_list) +{ + gdouble result; + + fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN, &result), + "Tag list contains track gain value (but should not)"); + fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK, &result), + "Tag list contains track peak value (but should not)"); +} + +static void +fail_if_album_tags (const GstTagList * tag_list) +{ + gdouble result; + + fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN, &result), + "Tag list contains album gain value (but should not)"); + fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK, &result), + "Tag list contains album peak value (but should not)"); +} + +static void +fail_unless_num_tracks (GstElement * element, guint num_tracks) +{ + guint current; + + g_object_get (element, "num-tracks", ¤t, NULL); + fail_unless (current == num_tracks, + "num-tracks property has incorrect value %u, expected %u", + current, num_tracks); +} + +/* Functions that create buffers with constant sample values, for peak + * tests. */ + +static GstBuffer * +test_buffer_const_float_mono (gint sample_rate, gsize n_frames, gfloat value) +{ + GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat)); + gfloat *data = (gfloat *) GST_BUFFER_DATA (buf); + GstCaps *caps; + gint i; + + for (i = n_frames; i--;) + *data++ = value; + + caps = gst_caps_new_simple ("audio/x-raw-float", + "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1, + "endianess", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL); + gst_buffer_set_caps (buf, caps); + gst_caps_unref (caps); + + ASSERT_BUFFER_REFCOUNT (buf, "buf", 1); + + return buf; +} + +static GstBuffer * +test_buffer_const_float_stereo (gint sample_rate, gsize n_frames, + gfloat value_l, gfloat value_r) +{ + GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat) * 2); + gfloat *data = (gfloat *) GST_BUFFER_DATA (buf); + GstCaps *caps; + gint i; + + for (i = n_frames; i--;) { + *data++ = value_l; + *data++ = value_r; + } + + caps = gst_caps_new_simple ("audio/x-raw-float", + "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2, + "endianess", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL); + gst_buffer_set_caps (buf, caps); + gst_caps_unref (caps); + + ASSERT_BUFFER_REFCOUNT (buf, "buf", 1); + + return buf; +} + +static GstBuffer * +test_buffer_const_int16_mono (gint sample_rate, gint depth, gsize n_frames, + gint16 value) +{ + GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16)); + gint16 *data = (gint16 *) GST_BUFFER_DATA (buf); + GstCaps *caps; + gint i; + + for (i = n_frames; i--;) + *data++ = value; + + caps = gst_caps_new_simple ("audio/x-raw-int", + "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1, + "endianess", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE, + "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL); + gst_buffer_set_caps (buf, caps); + gst_caps_unref (caps); + + ASSERT_BUFFER_REFCOUNT (buf, "buf", 1); + + return buf; +} + +static GstBuffer * +test_buffer_const_int16_stereo (gint sample_rate, gint depth, gsize n_frames, + gint16 value_l, gint16 value_r) +{ + GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16) * 2); + gint16 *data = (gint16 *) GST_BUFFER_DATA (buf); + GstCaps *caps; + gint i; + + for (i = n_frames; i--;) { + *data++ = value_l; + *data++ = value_r; + } + + caps = gst_caps_new_simple ("audio/x-raw-int", + "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2, + "endianess", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE, + "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL); + gst_buffer_set_caps (buf, caps); + gst_caps_unref (caps); + + ASSERT_BUFFER_REFCOUNT (buf, "buf", 1); + + return buf; +} + +/* Functions that create data buffers containing square signal + * waveforms. */ + +static GstBuffer * +test_buffer_square_float_mono (gint * accumulator, gint sample_rate, + gsize n_frames, gfloat value) +{ + GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat)); + gfloat *data = (gfloat *) GST_BUFFER_DATA (buf); + GstCaps *caps; + gint i; + + for (i = n_frames; i--;) { + *accumulator += 1; + *accumulator %= 96; + + if (*accumulator < 48) + *data++ = value; + else + *data++ = -value; + } + + caps = gst_caps_new_simple ("audio/x-raw-float", + "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1, + "endianess", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL); + gst_buffer_set_caps (buf, caps); + gst_caps_unref (caps); + + ASSERT_BUFFER_REFCOUNT (buf, "buf", 1); + + return buf; +} + +static GstBuffer * +test_buffer_square_float_stereo (gint * accumulator, gint sample_rate, + gsize n_frames, gfloat value_l, gfloat value_r) +{ + GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat) * 2); + gfloat *data = (gfloat *) GST_BUFFER_DATA (buf); + GstCaps *caps; + gint i; + + for (i = n_frames; i--;) { + *accumulator += 1; + *accumulator %= 96; + + if (*accumulator < 48) { + *data++ = value_l; + *data++ = value_r; + } else { + *data++ = -value_l; + *data++ = -value_r; + } + } + + caps = gst_caps_new_simple ("audio/x-raw-float", + "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2, + "endianess", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL); + gst_buffer_set_caps (buf, caps); + gst_caps_unref (caps); + + ASSERT_BUFFER_REFCOUNT (buf, "buf", 1); + + return buf; +} + +static GstBuffer * +test_buffer_square_int16_mono (gint * accumulator, gint sample_rate, + gint depth, gsize n_frames, gint16 value) +{ + GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16)); + gint16 *data = (gint16 *) GST_BUFFER_DATA (buf); + GstCaps *caps; + gint i; + + for (i = n_frames; i--;) { + *accumulator += 1; + *accumulator %= 96; + + if (*accumulator < 48) + *data++ = value; + else + *data++ = -MAX (value, -32767); + } + + caps = gst_caps_new_simple ("audio/x-raw-int", + "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1, + "endianess", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE, + "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL); + gst_buffer_set_caps (buf, caps); + gst_caps_unref (caps); + + ASSERT_BUFFER_REFCOUNT (buf, "buf", 1); + + return buf; +} + +static GstBuffer * +test_buffer_square_int16_stereo (gint * accumulator, gint sample_rate, + gint depth, gsize n_frames, gint16 value_l, gint16 value_r) +{ + GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16) * 2); + gint16 *data = (gint16 *) GST_BUFFER_DATA (buf); + GstCaps *caps; + gint i; + + for (i = n_frames; i--;) { + *accumulator += 1; + *accumulator %= 96; + + if (*accumulator < 48) { + *data++ = value_l; + *data++ = value_r; + } else { + *data++ = -MAX (value_l, -32767); + *data++ = -MAX (value_r, -32767); + } + } + + caps = gst_caps_new_simple ("audio/x-raw-int", + "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2, + "endianess", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE, + "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL); + gst_buffer_set_caps (buf, caps); + gst_caps_unref (caps); + + ASSERT_BUFFER_REFCOUNT (buf, "buf", 1); + + return buf; +} + +static void +push_buffer (GstBuffer * buf) +{ + /* gst_pad_push steals a reference. */ + fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK); + ASSERT_BUFFER_REFCOUNT (buf, "buf", 1); +} + +/*** Start of the tests. ***/ + +/* This test looks redundant, but early versions of the element + * crashed when doing, well, nothing: */ + +GST_START_TEST (test_no_buffer) +{ + GstElement *element = setup_rganalysis (); + + set_playing_state (element); + send_eos_event (element); + poll_eos (element); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +GST_START_TEST (test_no_buffer_album_1) +{ + GstElement *element = setup_rganalysis (); + + set_playing_state (element); + + /* Single track: */ + send_eos_event (element); + poll_eos (element); + + /* First album: */ + g_object_set (element, "num-tracks", 3, NULL); + + send_eos_event (element); + poll_eos (element); + fail_unless_num_tracks (element, 2); + + send_eos_event (element); + poll_eos (element); + fail_unless_num_tracks (element, 1); + + send_eos_event (element); + poll_eos (element); + fail_unless_num_tracks (element, 0); + + /* Second album: */ + g_object_set (element, "num-tracks", 2, NULL); + + send_eos_event (element); + poll_eos (element); + fail_unless_num_tracks (element, 1); + + send_eos_event (element); + poll_eos (element); + fail_unless_num_tracks (element, 0); + + /* Single track: */ + send_eos_event (element); + poll_eos (element); + fail_unless_num_tracks (element, 0); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +GST_START_TEST (test_no_buffer_album_2) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + gint accumulator = 0; + gint i; + + g_object_set (element, "num-tracks", 3, NULL); + set_playing_state (element); + + /* No buffer for the first track. */ + + send_eos_event (element); + /* No tags should be posted, there was nothing to analyze: */ + poll_eos (element); + fail_unless_num_tracks (element, 2); + + /* A test waveform with known gain result as second track: */ + + for (i = 20; i--;) + push_buffer (test_buffer_square_float_mono (&accumulator, 44100, 512, + 0.25)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, -6.20); + /* Album is not finished yet: */ + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 1); + + /* No buffer for the last track. */ + + send_eos_event (element); + + tag_list = poll_tags (element); + fail_unless_album_peak (tag_list, 0.25); + fail_unless_album_gain (tag_list, -6.20); + /* No track tags should be posted, as there was no data for it: */ + fail_if_track_tags (tag_list); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 0); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +GST_START_TEST (test_empty_buffers) +{ + GstElement *element = setup_rganalysis (); + + set_playing_state (element); + + /* Single track: */ + push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0)); + send_eos_event (element); + poll_eos (element); + + /* First album: */ + g_object_set (element, "num-tracks", 2, NULL); + + push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0)); + send_eos_event (element); + poll_eos (element); + fail_unless_num_tracks (element, 1); + + push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0)); + send_eos_event (element); + poll_eos (element); + fail_unless_num_tracks (element, 0); + + /* Second album, with a single track: */ + g_object_set (element, "num-tracks", 1, NULL); + push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0)); + send_eos_event (element); + poll_eos (element); + fail_unless_num_tracks (element, 0); + + /* Single track: */ + push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0)); + send_eos_event (element); + poll_eos (element); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +/* Tests for correctness of the peak values. */ + +/* Float peak test. For stereo, one channel has the constant value of + * -1.369, the other one 0.0. This tests many things: The result peak + * value should occur on any channel. The peak is of course the + * absolute amplitude, so 1.369 should be the result. This will also + * detect if the code uses the absolute value during the comparison. + * If it is buggy it will return 0.0 since 0.0 > -1.369. Furthermore, + * this makes sure that there is no problem with headroom (exceeding + * 0dBFS). In the wild you get float samples > 1.0 from stuff like + * vorbis. */ + +GST_START_TEST (test_peak_float) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + + set_playing_state (element); + push_buffer (test_buffer_const_float_stereo (8000, 512, -1.369, 0.0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.369); + gst_tag_list_free (tag_list); + + /* Swapped channels. */ + push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, -1.369)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.369); + gst_tag_list_free (tag_list); + + /* Mono. */ + push_buffer (test_buffer_const_float_mono (8000, 512, -1.369)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.369); + gst_tag_list_free (tag_list); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +GST_START_TEST (test_peak_int16_16) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + + set_playing_state (element); + + /* Half amplitude. */ + push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 1 << 14, 0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + /* Swapped channels. */ + push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, 1 << 14)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + /* Mono. */ + push_buffer (test_buffer_const_int16_mono (8000, 16, 512, 1 << 14)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + /* Half amplitude, negative variant. */ + push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -1 << 14, 0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + /* Swapped channels. */ + push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -1 << 14)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + /* Mono. */ + push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -1 << 14)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + + /* Now check for correct normalization of the peak value: Sample + * values of this format range from -32768 to 32767. So for the + * highest positive amplitude we do not reach 1.0, only for + * -32768! */ + + push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 32767, 0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 32767. / 32768.); + gst_tag_list_free (tag_list); + + /* Swapped channels. */ + push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, 32767)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 32767. / 32768.); + gst_tag_list_free (tag_list); + + /* Mono. */ + push_buffer (test_buffer_const_int16_mono (8000, 16, 512, 32767)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 32767. / 32768.); + gst_tag_list_free (tag_list); + + + /* Negative variant, reaching 1.0. */ + push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -32768, 0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.0); + gst_tag_list_free (tag_list); + + /* Swapped channels. */ + push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -32768)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.0); + gst_tag_list_free (tag_list); + + /* Mono. */ + push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -32768)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.0); + gst_tag_list_free (tag_list); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +/* Same as the test before, but with 8 bits (packed into 16 bits). */ + +GST_START_TEST (test_peak_int16_8) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + + set_playing_state (element); + + /* Half amplitude. */ + push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 1 << 6, 0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + /* Swapped channels. */ + push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, 1 << 6)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + /* Mono. */ + push_buffer (test_buffer_const_int16_mono (8000, 8, 512, 1 << 6)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + + /* Half amplitude, negative variant. */ + push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, -1 << 6, 0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + /* Swapped channels. */ + push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, -1 << 6)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + /* Mono. */ + push_buffer (test_buffer_const_int16_mono (8000, 8, 512, -1 << 6)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + + + /* Almost full amplitude (maximum positive value). */ + push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, (1 << 7) - 1, 0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.9921875); + gst_tag_list_free (tag_list); + + /* Swapped channels. */ + push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, (1 << 7) - 1)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.9921875); + gst_tag_list_free (tag_list); + + /* Mono. */ + push_buffer (test_buffer_const_int16_mono (8000, 8, 512, (1 << 7) - 1)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.9921875); + gst_tag_list_free (tag_list); + + + /* Full amplitude (maximum negative value). */ + push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, -1 << 7, 0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.0); + gst_tag_list_free (tag_list); + + /* Swapped channels. */ + push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, -1 << 7)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.0); + gst_tag_list_free (tag_list); + + /* Mono. */ + push_buffer (test_buffer_const_int16_mono (8000, 8, 512, -1 << 7)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.0); + gst_tag_list_free (tag_list); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +GST_START_TEST (test_peak_album) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + + g_object_set (element, "num-tracks", 2, NULL); + set_playing_state (element); + + push_buffer (test_buffer_const_float_stereo (8000, 1024, 1.0, 0.0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.0); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 1); + + push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.0, 0.5)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + fail_unless_album_peak (tag_list, 1.0); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 0); + + /* Try a second album: */ + g_object_set (element, "num-tracks", 3, NULL); + + push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.4, 0.4)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.4); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 2); + + push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.45, 0.45)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.45); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 1); + + push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.2, 0.2)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.2); + fail_unless_album_peak (tag_list, 0.45); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 0); + + /* And now a single track, not in album mode (num-tracks is 0 + * now): */ + push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.1, 0.1)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.1); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +/* Switching from track to album mode. */ + +GST_START_TEST (test_peak_track_album) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + guint num; + + set_playing_state (element); + + push_buffer (test_buffer_const_float_mono (8000, 1024, 1.0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.0); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + + g_object_set (element, "num-tracks", 1, NULL); + push_buffer (test_buffer_const_float_mono (8000, 1024, 0.5)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + fail_unless_album_peak (tag_list, 0.5); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 0); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +/* Disabling album processing before the end of the album. Probably a + * rare edge case and applications should not rely on this to work. + * They need to send the element to the READY state to clear up after + * an aborted album anyway since they might need to process another + * album afterwards. */ + +GST_START_TEST (test_peak_album_abort_to_track) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + guint num; + + g_object_set (element, "num-tracks", 2, NULL); + set_playing_state (element); + + push_buffer (test_buffer_const_float_stereo (8000, 1024, 1.0, 0.0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 1.0); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 1); + + g_object_set (element, "num-tracks", 0, NULL); + + push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.0, 0.5)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +GST_START_TEST (test_gain_album) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + gint accumulator; + gint i; + + g_object_set (element, "num-tracks", 3, NULL); + set_playing_state (element); + + /* The three tracks are constructed such that if any of these is in + * fact ignored for the album gain, the album gain will differ. */ + + accumulator = 0; + for (i = 8; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512, + 0.75, 0.75)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.75); + fail_unless_track_gain (tag_list, -15.70); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + + accumulator = 0; + for (i = 12; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512, + 0.5, 0.5)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.5); + fail_unless_track_gain (tag_list, -12.22); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + + accumulator = 0; + for (i = 180; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512, + 0.25, 0.25)); + send_eos_event (element); + + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, -6.20); + fail_unless_album_peak (tag_list, 0.75); + /* Strangely, wavegain reports -12.17 for the album, but the fixed + * metaflac agrees to us. Could be a 32767 vs. 32768 issue. */ + fail_unless_album_gain (tag_list, -12.18); + gst_tag_list_free (tag_list); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +/* Checks ensuring that the "forced" property works as advertised. */ + +GST_START_TEST (test_forced) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + gint accumulator = 0; + gint i; + + g_object_set (element, "forced", FALSE, NULL); + set_playing_state (element); + + tag_list = gst_tag_list_new (); + /* Provided values are totally arbitrary. */ + gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, + GST_TAG_TRACK_PEAK, 1.0, GST_TAG_TRACK_GAIN, 2.21, NULL); + send_tag_event (element, tag_list); + + for (i = 20; i--;) + push_buffer (test_buffer_const_float_stereo (44100, 512, 0.5, 0.5)); + send_eos_event (element); + /* This fails if a tag message is generated: */ + poll_eos (element); + + /* Now back to a track without tags. */ + + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512, + 0.25, 0.25)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, get_expected_gain (44100)); + gst_tag_list_free (tag_list); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +/* Sending track gain and peak in separate tag lists. */ + +GST_START_TEST (test_forced_separate) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + gint accumulator = 0; + gint i; + + g_object_set (element, "forced", FALSE, NULL); + set_playing_state (element); + + tag_list = gst_tag_list_new (); + gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, GST_TAG_TRACK_GAIN, 2.21, + NULL); + send_tag_event (element, tag_list); + + tag_list = gst_tag_list_new (); + gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, GST_TAG_TRACK_PEAK, 1.0, + NULL); + send_tag_event (element, tag_list); + + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512, + 0.5, 0.5)); + send_eos_event (element); + /* This fails if a tag message is generated: */ + poll_eos (element); + + /* Now a track without tags. */ + + accumulator = 0; + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512, + 0.25, 0.25)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, get_expected_gain (44100)); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +/* A TAG event is sent _after_ data has already been processed. In + * real pipelines, this could happen if there is more than one + * rganalysis element (by accident). While it would have analyzed all + * the data prior to receiving the event, I expect it to not post its + * results if not forced. This test is almost equivalent to + * test_forced. */ + +GST_START_TEST (test_forced_after_data) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + gint accumulator = 0; + gint i; + + g_object_set (element, "forced", FALSE, NULL); + set_playing_state (element); + + for (i = 20; i--;) + push_buffer (test_buffer_const_float_stereo (8000, 512, 0.5, 0.5)); + + tag_list = gst_tag_list_new (); + gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, + GST_TAG_TRACK_PEAK, 1.0, GST_TAG_TRACK_GAIN, 2.21, NULL); + send_tag_event (element, tag_list); + + send_eos_event (element); + poll_eos (element); + + /* Now back to a normal track, this one has no tags: */ + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25, + 0.25)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, get_expected_gain (8000)); + gst_tag_list_free (tag_list); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +/* Like test_forced, but *analyze* an album afterwards. The two tests + * following this one check the *skipping* of albums. */ + +GST_START_TEST (test_forced_album) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + gint accumulator; + gint i; + + g_object_set (element, "forced", FALSE, NULL); + set_playing_state (element); + + tag_list = gst_tag_list_new (); + /* Provided values are totally arbitrary. */ + gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, + GST_TAG_TRACK_PEAK, 1.0, GST_TAG_TRACK_GAIN, 2.21, NULL); + send_tag_event (element, tag_list); + + accumulator = 0; + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512, + 0.5, 0.5)); + send_eos_event (element); + /* This fails if a tag message is generated: */ + poll_eos (element); + + /* Now an album without tags. */ + g_object_set (element, "num-tracks", 2, NULL); + + accumulator = 0; + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512, + 0.25, 0.25)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, get_expected_gain (44100)); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 1); + + accumulator = 0; + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512, + 0.25, 0.25)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, get_expected_gain (44100)); + fail_unless_album_peak (tag_list, 0.25); + fail_unless_album_gain (tag_list, get_expected_gain (44100)); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 0); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +GST_START_TEST (test_forced_album_skip) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + gint accumulator = 0; + gint i; + + g_object_set (element, "forced", FALSE, "num-tracks", 2, NULL); + set_playing_state (element); + + tag_list = gst_tag_list_new (); + /* Provided values are totally arbitrary. */ + gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, + GST_TAG_TRACK_PEAK, 0.75, GST_TAG_TRACK_GAIN, 2.21, + GST_TAG_ALBUM_PEAK, 0.80, GST_TAG_ALBUM_GAIN, -0.11, NULL); + send_tag_event (element, tag_list); + + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25, + 0.25)); + send_eos_event (element); + poll_eos (element); + fail_unless_num_tracks (element, 1); + + /* This track has no tags, but needs to be skipped anyways since we + * are in album processing mode. */ + for (i = 20; i--;) + push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, 0.0)); + send_eos_event (element); + poll_eos (element); + fail_unless_num_tracks (element, 0); + + /* Normal track after the album. Of course not to be skipped. */ + accumulator = 0; + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25, + 0.25)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, get_expected_gain (8000)); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +GST_START_TEST (test_forced_album_no_skip) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + gint accumulator = 0; + gint i; + + g_object_set (element, "forced", FALSE, "num-tracks", 2, NULL); + set_playing_state (element); + + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25, + 0.25)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, get_expected_gain (8000)); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 1); + + /* The second track has indeed full tags, but although being not + * forced, this one has to be processed because album processing is + * on. */ + tag_list = gst_tag_list_new (); + /* Provided values are totally arbitrary. */ + gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, + GST_TAG_TRACK_PEAK, 0.75, GST_TAG_TRACK_GAIN, 2.21, + GST_TAG_ALBUM_PEAK, 0.80, GST_TAG_ALBUM_GAIN, -0.11, NULL); + send_tag_event (element, tag_list); + for (i = 20; i--;) + push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, 0.0)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.0); + fail_unless_track_gain (tag_list, SILENCE_GAIN); + /* Second track was just silence so the album peak equals the first + * track's peak. */ + fail_unless_album_peak (tag_list, 0.25); + /* Statistical processing leads to the second track being + * ignored for the gain (because it is so short): */ + fail_unless_album_gain (tag_list, get_expected_gain (8000)); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 0); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +GST_START_TEST (test_forced_abort_album_no_skip) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + gint accumulator = 0; + gint i; + + g_object_set (element, "forced", FALSE, "num-tracks", 2, NULL); + set_playing_state (element); + + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25, + 0.25)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, get_expected_gain (8000)); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + fail_unless_num_tracks (element, 1); + + /* Disabling album processing before end of album: */ + g_object_set (element, "num-tracks", 0, NULL); + + /* Processing a track that has to be skipped. */ + tag_list = gst_tag_list_new (); + /* Provided values are totally arbitrary. */ + gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, + GST_TAG_TRACK_PEAK, 0.75, GST_TAG_TRACK_GAIN, 2.21, + GST_TAG_ALBUM_PEAK, 0.80, GST_TAG_ALBUM_GAIN, -0.11, NULL); + send_tag_event (element, tag_list); + for (i = 20; i--;) + push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, 0.0)); + send_eos_event (element); + poll_eos (element); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +GST_START_TEST (test_reference_level) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + gint accumulator = 0; + gint i; + + g_object_set (element, "reference-level", 83., "num-tracks", 2, NULL); + set_playing_state (element); + + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512, + 0.25, 0.25)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, get_expected_gain (44100) - 6.); + fail_if_album_tags (tag_list); + gst_tag_list_free (tag_list); + + accumulator = 0; + for (i = 20; i--;) + push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512, + 0.25, 0.25)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, get_expected_gain (44100) - 6.); + fail_unless_album_peak (tag_list, 0.25); + /* We provided the same waveform twice, with a reset separating + * them. Therefore, the album gain matches the track gain. */ + fail_unless_album_gain (tag_list, get_expected_gain (44100) - 6.); + gst_tag_list_free (tag_list); + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +GST_START_TEST (test_all_formats) +{ + GstElement *element = setup_rganalysis (); + GstTagList *tag_list; + gint accumulator = 0; + gint i, j; + + set_playing_state (element); + for (i = G_N_ELEMENTS (supported_rates); i--;) { + accumulator = 0; + for (j = 0; j < 4; j++) + push_buffer (test_buffer_square_float_stereo (&accumulator, + supported_rates[i].sample_rate, 512, 0.25, 0.25)); + for (j = 0; j < 3; j++) + push_buffer (test_buffer_square_float_mono (&accumulator, + supported_rates[i].sample_rate, 512, 0.25)); + for (j = 0; j < 4; j++) + push_buffer (test_buffer_square_int16_stereo (&accumulator, + supported_rates[i].sample_rate, 16, 512, 1 << 13, 1 << 13)); + for (j = 0; j < 3; j++) + push_buffer (test_buffer_square_int16_mono (&accumulator, + supported_rates[i].sample_rate, 16, 512, 1 << 13)); + for (j = 0; j < 3; j++) + push_buffer (test_buffer_square_int16_stereo (&accumulator, + supported_rates[i].sample_rate, 8, 512, 1 << 5, 1 << 5)); + for (j = 0; j < 3; j++) + push_buffer (test_buffer_square_int16_mono (&accumulator, + supported_rates[i].sample_rate, 8, 512, 1 << 5)); + send_eos_event (element); + tag_list = poll_tags (element); + fail_unless_track_peak (tag_list, 0.25); + fail_unless_track_gain (tag_list, supported_rates[i].gain); + gst_tag_list_free (tag_list); + } + + cleanup_rganalysis (element); +} + +GST_END_TEST; + +/* Checks ensuring all advertised supported sample rates are really + * accepted, for integer and float, mono and stereo. This also + * verifies that the correct gain is computed for all formats (except + * odd bit depths). */ + +#define MAKE_GAIN_TEST_FLOAT_MONO(sample_rate) \ + GST_START_TEST (test_gain_float_mono_##sample_rate) \ +{ \ + GstElement *element = setup_rganalysis (); \ + GstTagList *tag_list; \ + gint accumulator = 0; \ + gint i; \ + \ + set_playing_state (element); \ + \ + for (i = 0; i < 20; i++) \ + push_buffer (test_buffer_square_float_mono (&accumulator, \ + sample_rate, 512, 0.25)); \ + send_eos_event (element); \ + tag_list = poll_tags (element); \ + fail_unless_track_peak (tag_list, 0.25); \ + fail_unless_track_gain (tag_list, \ + get_expected_gain (sample_rate)); \ + gst_tag_list_free (tag_list); \ + \ + cleanup_rganalysis (element); \ +} \ + \ +GST_END_TEST; + +#define MAKE_GAIN_TEST_FLOAT_STEREO(sample_rate) \ + GST_START_TEST (test_gain_float_stereo_##sample_rate) \ +{ \ + GstElement *element = setup_rganalysis (); \ + GstTagList *tag_list; \ + gint accumulator = 0; \ + gint i; \ + \ + set_playing_state (element); \ + \ + for (i = 0; i < 20; i++) \ + push_buffer (test_buffer_square_float_stereo (&accumulator, \ + sample_rate, 512, 0.25, 0.25)); \ + send_eos_event (element); \ + tag_list = poll_tags (element); \ + fail_unless_track_peak (tag_list, 0.25); \ + fail_unless_track_gain (tag_list, \ + get_expected_gain (sample_rate)); \ + gst_tag_list_free (tag_list); \ + \ + cleanup_rganalysis (element); \ +} \ + \ +GST_END_TEST; + +#define MAKE_GAIN_TEST_INT16_MONO(sample_rate, depth) \ + GST_START_TEST (test_gain_int16_##depth##_mono_##sample_rate) \ +{ \ + GstElement *element = setup_rganalysis (); \ + GstTagList *tag_list; \ + gint accumulator = 0; \ + gint i; \ + \ + set_playing_state (element); \ + \ + for (i = 0; i < 20; i++) \ + push_buffer (test_buffer_square_int16_mono (&accumulator, \ + sample_rate, depth, 512, 1 << (13 + depth - 16))); \ + \ + send_eos_event (element); \ + tag_list = poll_tags (element); \ + fail_unless_track_peak (tag_list, 0.25); \ + fail_unless_track_gain (tag_list, \ + get_expected_gain (sample_rate)); \ + gst_tag_list_free (tag_list); \ + \ + cleanup_rganalysis (element); \ +} \ + \ +GST_END_TEST; + +#define MAKE_GAIN_TEST_INT16_STEREO(sample_rate, depth) \ + GST_START_TEST (test_gain_int16_##depth##_stereo_##sample_rate) \ +{ \ + GstElement *element = setup_rganalysis (); \ + GstTagList *tag_list; \ + gint accumulator = 0; \ + gint i; \ + \ + set_playing_state (element); \ + \ + for (i = 0; i < 20; i++) \ + push_buffer (test_buffer_square_int16_stereo (&accumulator, \ + sample_rate, depth, 512, 1 << (13 + depth - 16), \ + 1 << (13 + depth - 16))); \ + send_eos_event (element); \ + tag_list = poll_tags (element); \ + fail_unless_track_peak (tag_list, 0.25); \ + fail_unless_track_gain (tag_list, \ + get_expected_gain (sample_rate)); \ + gst_tag_list_free (tag_list); \ + \ + cleanup_rganalysis (element); \ +} \ + \ +GST_END_TEST; + +MAKE_GAIN_TEST_FLOAT_MONO (8000); +MAKE_GAIN_TEST_FLOAT_MONO (11025); +MAKE_GAIN_TEST_FLOAT_MONO (12000); +MAKE_GAIN_TEST_FLOAT_MONO (16000); +MAKE_GAIN_TEST_FLOAT_MONO (22050); +MAKE_GAIN_TEST_FLOAT_MONO (24000); +MAKE_GAIN_TEST_FLOAT_MONO (32000); +MAKE_GAIN_TEST_FLOAT_MONO (44100); +MAKE_GAIN_TEST_FLOAT_MONO (48000); + +MAKE_GAIN_TEST_FLOAT_STEREO (8000); +MAKE_GAIN_TEST_FLOAT_STEREO (11025); +MAKE_GAIN_TEST_FLOAT_STEREO (12000); +MAKE_GAIN_TEST_FLOAT_STEREO (16000); +MAKE_GAIN_TEST_FLOAT_STEREO (22050); +MAKE_GAIN_TEST_FLOAT_STEREO (24000); +MAKE_GAIN_TEST_FLOAT_STEREO (32000); +MAKE_GAIN_TEST_FLOAT_STEREO (44100); +MAKE_GAIN_TEST_FLOAT_STEREO (48000); + +MAKE_GAIN_TEST_INT16_MONO (8000, 16); +MAKE_GAIN_TEST_INT16_MONO (11025, 16); +MAKE_GAIN_TEST_INT16_MONO (12000, 16); +MAKE_GAIN_TEST_INT16_MONO (16000, 16); +MAKE_GAIN_TEST_INT16_MONO (22050, 16); +MAKE_GAIN_TEST_INT16_MONO (24000, 16); +MAKE_GAIN_TEST_INT16_MONO (32000, 16); +MAKE_GAIN_TEST_INT16_MONO (44100, 16); +MAKE_GAIN_TEST_INT16_MONO (48000, 16); + +MAKE_GAIN_TEST_INT16_STEREO (8000, 16); +MAKE_GAIN_TEST_INT16_STEREO (11025, 16); +MAKE_GAIN_TEST_INT16_STEREO (12000, 16); +MAKE_GAIN_TEST_INT16_STEREO (16000, 16); +MAKE_GAIN_TEST_INT16_STEREO (22050, 16); +MAKE_GAIN_TEST_INT16_STEREO (24000, 16); +MAKE_GAIN_TEST_INT16_STEREO (32000, 16); +MAKE_GAIN_TEST_INT16_STEREO (44100, 16); +MAKE_GAIN_TEST_INT16_STEREO (48000, 16); + +MAKE_GAIN_TEST_INT16_MONO (8000, 8); +MAKE_GAIN_TEST_INT16_MONO (11025, 8); +MAKE_GAIN_TEST_INT16_MONO (12000, 8); +MAKE_GAIN_TEST_INT16_MONO (16000, 8); +MAKE_GAIN_TEST_INT16_MONO (22050, 8); +MAKE_GAIN_TEST_INT16_MONO (24000, 8); +MAKE_GAIN_TEST_INT16_MONO (32000, 8); +MAKE_GAIN_TEST_INT16_MONO (44100, 8); +MAKE_GAIN_TEST_INT16_MONO (48000, 8); + +MAKE_GAIN_TEST_INT16_STEREO (8000, 8); +MAKE_GAIN_TEST_INT16_STEREO (11025, 8); +MAKE_GAIN_TEST_INT16_STEREO (12000, 8); +MAKE_GAIN_TEST_INT16_STEREO (16000, 8); +MAKE_GAIN_TEST_INT16_STEREO (22050, 8); +MAKE_GAIN_TEST_INT16_STEREO (24000, 8); +MAKE_GAIN_TEST_INT16_STEREO (32000, 8); +MAKE_GAIN_TEST_INT16_STEREO (44100, 8); +MAKE_GAIN_TEST_INT16_STEREO (48000, 8); + +Suite * +rganalysis_suite (void) +{ + Suite *s = suite_create ("rganalysis"); + TCase *tc_chain = tcase_create ("general"); + + suite_add_tcase (s, tc_chain); + + tcase_add_test (tc_chain, test_no_buffer); + tcase_add_test (tc_chain, test_no_buffer_album_1); + tcase_add_test (tc_chain, test_no_buffer_album_2); + tcase_add_test (tc_chain, test_empty_buffers); + + tcase_add_test (tc_chain, test_peak_float); + tcase_add_test (tc_chain, test_peak_int16_16); + tcase_add_test (tc_chain, test_peak_int16_8); + + tcase_add_test (tc_chain, test_peak_album); + tcase_add_test (tc_chain, test_peak_track_album); + tcase_add_test (tc_chain, test_peak_album_abort_to_track); + + tcase_add_test (tc_chain, test_gain_album); + + tcase_add_test (tc_chain, test_forced); + tcase_add_test (tc_chain, test_forced_separate); + tcase_add_test (tc_chain, test_forced_after_data); + tcase_add_test (tc_chain, test_forced_album); + tcase_add_test (tc_chain, test_forced_album_skip); + tcase_add_test (tc_chain, test_forced_album_no_skip); + tcase_add_test (tc_chain, test_forced_abort_album_no_skip); + + tcase_add_test (tc_chain, test_reference_level); + + tcase_add_test (tc_chain, test_all_formats); + + tcase_add_test (tc_chain, test_gain_float_mono_8000); + tcase_add_test (tc_chain, test_gain_float_mono_11025); + tcase_add_test (tc_chain, test_gain_float_mono_12000); + tcase_add_test (tc_chain, test_gain_float_mono_16000); + tcase_add_test (tc_chain, test_gain_float_mono_22050); + tcase_add_test (tc_chain, test_gain_float_mono_24000); + tcase_add_test (tc_chain, test_gain_float_mono_32000); + tcase_add_test (tc_chain, test_gain_float_mono_44100); + tcase_add_test (tc_chain, test_gain_float_mono_48000); + + tcase_add_test (tc_chain, test_gain_float_stereo_8000); + tcase_add_test (tc_chain, test_gain_float_stereo_11025); + tcase_add_test (tc_chain, test_gain_float_stereo_12000); + tcase_add_test (tc_chain, test_gain_float_stereo_16000); + tcase_add_test (tc_chain, test_gain_float_stereo_22050); + tcase_add_test (tc_chain, test_gain_float_stereo_24000); + tcase_add_test (tc_chain, test_gain_float_stereo_32000); + tcase_add_test (tc_chain, test_gain_float_stereo_44100); + tcase_add_test (tc_chain, test_gain_float_stereo_48000); + + tcase_add_test (tc_chain, test_gain_int16_16_mono_8000); + tcase_add_test (tc_chain, test_gain_int16_16_mono_11025); + tcase_add_test (tc_chain, test_gain_int16_16_mono_12000); + tcase_add_test (tc_chain, test_gain_int16_16_mono_16000); + tcase_add_test (tc_chain, test_gain_int16_16_mono_22050); + tcase_add_test (tc_chain, test_gain_int16_16_mono_24000); + tcase_add_test (tc_chain, test_gain_int16_16_mono_32000); + tcase_add_test (tc_chain, test_gain_int16_16_mono_44100); + tcase_add_test (tc_chain, test_gain_int16_16_mono_48000); + + tcase_add_test (tc_chain, test_gain_int16_16_stereo_8000); + tcase_add_test (tc_chain, test_gain_int16_16_stereo_11025); + tcase_add_test (tc_chain, test_gain_int16_16_stereo_12000); + tcase_add_test (tc_chain, test_gain_int16_16_stereo_16000); + tcase_add_test (tc_chain, test_gain_int16_16_stereo_22050); + tcase_add_test (tc_chain, test_gain_int16_16_stereo_24000); + tcase_add_test (tc_chain, test_gain_int16_16_stereo_32000); + tcase_add_test (tc_chain, test_gain_int16_16_stereo_44100); + tcase_add_test (tc_chain, test_gain_int16_16_stereo_48000); + + tcase_add_test (tc_chain, test_gain_int16_8_mono_8000); + tcase_add_test (tc_chain, test_gain_int16_8_mono_11025); + tcase_add_test (tc_chain, test_gain_int16_8_mono_12000); + tcase_add_test (tc_chain, test_gain_int16_8_mono_16000); + tcase_add_test (tc_chain, test_gain_int16_8_mono_22050); + tcase_add_test (tc_chain, test_gain_int16_8_mono_24000); + tcase_add_test (tc_chain, test_gain_int16_8_mono_32000); + tcase_add_test (tc_chain, test_gain_int16_8_mono_44100); + tcase_add_test (tc_chain, test_gain_int16_8_mono_48000); + + tcase_add_test (tc_chain, test_gain_int16_8_stereo_8000); + tcase_add_test (tc_chain, test_gain_int16_8_stereo_11025); + tcase_add_test (tc_chain, test_gain_int16_8_stereo_12000); + tcase_add_test (tc_chain, test_gain_int16_8_stereo_16000); + tcase_add_test (tc_chain, test_gain_int16_8_stereo_22050); + tcase_add_test (tc_chain, test_gain_int16_8_stereo_24000); + tcase_add_test (tc_chain, test_gain_int16_8_stereo_32000); + tcase_add_test (tc_chain, test_gain_int16_8_stereo_44100); + tcase_add_test (tc_chain, test_gain_int16_8_stereo_48000); + + return s; +} + +int +main (int argc, char **argv) +{ + gint nf; + + Suite *s = rganalysis_suite (); + SRunner *sr = srunner_create (s); + + gst_check_init (&argc, &argv); + + srunner_run_all (sr, CK_ENV); + nf = srunner_ntests_failed (sr); + srunner_free (sr); + + return nf; +} -- 2.7.4