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authorJan Schmidt <thaytan@mad.scientist.com>2008-07-19 00:58:49 +0000
committerJan Schmidt <thaytan@mad.scientist.com>2008-07-19 00:58:49 +0000
commite985585a4ec8ec1a681c9643f6727a230fb536d7 (patch)
tree1b1fc2eeabad64ca5ba42b51cf42acc082686189 /gst
parent26cb95316c8043e05365337660c1e07b067f298e (diff)
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Remove interleave and replaygain plugins that have moved to -good
Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-bad-plugins-docs.sgml: * docs/plugins/gst-plugins-bad-plugins-sections.txt: * docs/plugins/gst-plugins-bad-plugins.args: * docs/plugins/gst-plugins-bad-plugins.hierarchy: * docs/plugins/gst-plugins-bad-plugins.interfaces: * docs/plugins/gst-plugins-bad-plugins.prerequisites: * docs/plugins/inspect/plugin-interleave.xml: * docs/plugins/inspect/plugin-replaygain.xml: * gst/interleave/Makefile.am: * gst/interleave/deinterleave.c: * gst/interleave/deinterleave.h: * gst/interleave/interleave.c: * gst/interleave/interleave.h: * gst/interleave/plugin.c: * gst/interleave/plugin.h: * gst/replaygain/Makefile.am: * gst/replaygain/gstrganalysis.c: * gst/replaygain/gstrganalysis.h: * gst/replaygain/gstrglimiter.c: * gst/replaygain/gstrglimiter.h: * gst/replaygain/gstrgvolume.c: * gst/replaygain/gstrgvolume.h: * gst/replaygain/replaygain.c: * gst/replaygain/replaygain.h: * gst/replaygain/rganalysis.c: * gst/replaygain/rganalysis.h: * tests/check/Makefile.am: * tests/check/elements/deinterleave.c: * tests/check/elements/interleave.c: * tests/check/elements/rganalysis.c: * tests/check/elements/rglimiter.c: * tests/check/elements/rgvolume.c: Remove interleave and replaygain plugins that have moved to -good
Diffstat (limited to 'gst')
-rw-r--r--gst/interleave/Makefile.am9
-rw-r--r--gst/interleave/deinterleave.c889
-rw-r--r--gst/interleave/deinterleave.h75
-rw-r--r--gst/interleave/interleave.c1352
-rw-r--r--gst/interleave/interleave.h89
-rw-r--r--gst/interleave/plugin.c44
-rw-r--r--gst/interleave/plugin.h31
-rw-r--r--gst/replaygain/Makefile.am21
-rw-r--r--gst/replaygain/gstrganalysis.c692
-rw-r--r--gst/replaygain/gstrganalysis.h85
-rw-r--r--gst/replaygain/gstrglimiter.c202
-rw-r--r--gst/replaygain/gstrglimiter.h64
-rw-r--r--gst/replaygain/gstrgvolume.c698
-rw-r--r--gst/replaygain/gstrgvolume.h88
-rw-r--r--gst/replaygain/replaygain.c53
-rw-r--r--gst/replaygain/replaygain.h36
-rw-r--r--gst/replaygain/rganalysis.c777
-rw-r--r--gst/replaygain/rganalysis.h56
18 files changed, 0 insertions, 5261 deletions
diff --git a/gst/interleave/Makefile.am b/gst/interleave/Makefile.am
deleted file mode 100644
index 3477933c..00000000
--- a/gst/interleave/Makefile.am
+++ /dev/null
@@ -1,9 +0,0 @@
-
-plugin_LTLIBRARIES = libgstinterleave.la
-
-libgstinterleave_la_SOURCES = plugin.c interleave.c deinterleave.c
-libgstinterleave_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
-libgstinterleave_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR)
-libgstinterleave_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
-
-noinst_HEADERS = plugin.h interleave.h deinterleave.h
diff --git a/gst/interleave/deinterleave.c b/gst/interleave/deinterleave.c
deleted file mode 100644
index 4c81d39d..00000000
--- a/gst/interleave/deinterleave.c
+++ /dev/null
@@ -1,889 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- * 2000 Wim Taymans <wtay@chello.be>
- * 2005 Wim Taymans <wim@fluendo.com>
- * 2007 Andy Wingo <wingo at pobox.com>
- * 2008 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * deinterleave.c: deinterleave samples
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 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
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/* TODO:
- * - handle changes in number of channels
- * - handle changes in channel positions
- * - better capsnego by using a buffer alloc function
- * and passing downstream caps changes upstream there
- */
-
-/**
- * SECTION:element-deinterleave
- * @see_also: interleave
- *
- * Splits one interleaved multichannel audio stream into many mono audio streams.
- *
- * This element handles all raw audio formats and supports changing the input caps as long as
- * all downstream elements can handle the new caps and the number of channels and the channel
- * positions stay the same. This restriction will be removed in later versions by adding or
- * removing some source pads as required.
- *
- * In most cases a queue and an audioconvert element should be added after each source pad
- * before further processing of the audio data.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch-0.10 filesrc location=/path/to/file.mp3 ! decodebin ! audioconvert ! "audio/x-raw-int,channels=2 ! deinterleave name=d d.src0 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel1.ogg d.src1 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel2.ogg
- * ]| Decodes an MP3 file and encodes the left and right channel into separate
- * Ogg Vorbis files.
- * |[
- * gst-launch-0.10 filesrc location=file.mp3 ! decodebin ! audioconvert ! "audio/x-raw-int,channels=2" ! deinterleave name=d interleave name=i ! audioconvert ! wavenc ! filesink location=test.wav d.src0 ! queue ! audioconvert ! i.sink1 d.src1 ! queue ! audioconvert ! i.sink0
- * ]| Decodes and deinterleaves a Stereo MP3 file into separate channels and
- * then interleaves the channels again to a WAV file with the channel with the
- * channels exchanged.
- * </refsect2>
- */
-
-#ifdef HAVE_CONFIG_H
-# include "config.h"
-#endif
-
-#include <gst/gst.h>
-#include <string.h>
-#include "deinterleave.h"
-
-GST_DEBUG_CATEGORY_STATIC (gst_deinterleave_debug);
-#define GST_CAT_DEFAULT gst_deinterleave_debug
-
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src%d",
- GST_PAD_SRC,
- GST_PAD_SOMETIMES,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) 1, "
- "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
- "width = (int) { 8, 16, 24, 32 }, "
- "depth = (int) [ 1, 32 ], "
- "signed = (boolean) { true, false }; "
- "audio/x-raw-float, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) 1, "
- "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, "
- "width = (int) { 32, 64 }")
- );
-
-static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, MAX ], "
- "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
- "width = (int) { 8, 16, 24, 32 }, "
- "depth = (int) [ 1, 32 ], "
- "signed = (boolean) { true, false }; "
- "audio/x-raw-float, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, MAX ], "
- "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, "
- "width = (int) { 32, 64 }")
- );
-
-#define MAKE_FUNC(type) \
-static void deinterleave_##type (guint##type *out, guint##type *in, \
- guint stride, guint nframes) \
-{ \
- gint i; \
- \
- for (i = 0; i < nframes; i++) { \
- out[i] = *in; \
- in += stride; \
- } \
-}
-
-MAKE_FUNC (8);
-MAKE_FUNC (16);
-MAKE_FUNC (32);
-MAKE_FUNC (64);
-
-static void
-deinterleave_24 (guint8 * out, guint8 * in, guint stride, guint nframes)
-{
- gint i;
-
- for (i = 0; i < nframes; i++) {
- memcpy (out, in, 3);
- out += 3;
- in += stride * 3;
- }
-}
-
-GST_BOILERPLATE (GstDeinterleave, gst_deinterleave, GstElement,
- GST_TYPE_ELEMENT);
-
-enum
-{
- PROP_0,
- PROP_KEEP_POSITIONS
-};
-
-static GstFlowReturn gst_deinterleave_chain (GstPad * pad, GstBuffer * buffer);
-
-static gboolean gst_deinterleave_sink_setcaps (GstPad * pad, GstCaps * caps);
-
-static GstCaps *gst_deinterleave_sink_getcaps (GstPad * pad);
-
-static gboolean gst_deinterleave_sink_activate_push (GstPad * pad,
- gboolean active);
-static gboolean gst_deinterleave_sink_event (GstPad * pad, GstEvent * event);
-
-static gboolean gst_deinterleave_src_query (GstPad * pad, GstQuery * query);
-
-static void gst_deinterleave_set_property (GObject * object,
- guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_deinterleave_get_property (GObject * object,
- guint prop_id, GValue * value, GParamSpec * pspec);
-
-
-static void
-gst_deinterleave_finalize (GObject * obj)
-{
- GstDeinterleave *self = GST_DEINTERLEAVE (obj);
-
- if (self->pos) {
- g_free (self->pos);
- self->pos = NULL;
- }
-
- if (self->pending_events) {
- g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref, NULL);
- g_list_free (self->pending_events);
- self->pending_events = NULL;
- }
-
- G_OBJECT_CLASS (parent_class)->finalize (obj);
-}
-
-static void
-gst_deinterleave_base_init (gpointer g_class)
-{
- GstElementClass *gstelement_class = (GstElementClass *) g_class;
-
- gst_element_class_set_details_simple (gstelement_class, "Audio deinterleaver",
- "Filter/Converter/Audio",
- "Splits one interleaved multichannel audio stream into many mono audio streams",
- "Andy Wingo <wingo at pobox.com>, "
- "Iain <iain@prettypeople.org>, "
- "Sebastian Dröge <slomo@circular-chaos.org>");
-
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&sink_template));
- gst_element_class_add_pad_template (gstelement_class,
- gst_static_pad_template_get (&src_template));
-}
-
-static void
-gst_deinterleave_class_init (GstDeinterleaveClass * klass)
-{
- GObjectClass *gobject_class = (GObjectClass *) klass;
-
- GST_DEBUG_CATEGORY_INIT (gst_deinterleave_debug, "deinterleave", 0,
- "deinterleave element");
-
- gobject_class->finalize = gst_deinterleave_finalize;
- gobject_class->set_property = gst_deinterleave_set_property;
- gobject_class->get_property = gst_deinterleave_get_property;
-
- /**
- * GstDeinterleave:keep-positions
- *
- * Keep positions: When enable the caps on the output buffers will
- * contain the original channel positions. This can be used to correctly
- * interleave the output again later but can also lead to unwanted effects
- * if the output should be handled as Mono.
- *
- */
- g_object_class_install_property (gobject_class, PROP_KEEP_POSITIONS,
- g_param_spec_boolean ("keep-positions", "Keep positions",
- "Keep the original channel positions on the output buffers",
- FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-}
-
-static void
-gst_deinterleave_init (GstDeinterleave * self, GstDeinterleaveClass * klass)
-{
- self->channels = 0;
- self->pos = NULL;
- self->keep_positions = FALSE;
- self->width = 0;
- self->func = NULL;
-
- /* Add sink pad */
- self->sink = gst_pad_new_from_static_template (&sink_template, "sink");
- gst_pad_set_chain_function (self->sink,
- GST_DEBUG_FUNCPTR (gst_deinterleave_chain));
- gst_pad_set_setcaps_function (self->sink,
- GST_DEBUG_FUNCPTR (gst_deinterleave_sink_setcaps));
- gst_pad_set_getcaps_function (self->sink,
- GST_DEBUG_FUNCPTR (gst_deinterleave_sink_getcaps));
- gst_pad_set_activatepush_function (self->sink,
- GST_DEBUG_FUNCPTR (gst_deinterleave_sink_activate_push));
- gst_pad_set_event_function (self->sink,
- GST_DEBUG_FUNCPTR (gst_deinterleave_sink_event));
- gst_element_add_pad (GST_ELEMENT (self), self->sink);
-}
-
-static void
-gst_deinterleave_add_new_pads (GstDeinterleave * self, GstCaps * caps)
-{
- GstPad *pad;
-
- guint i;
-
- for (i = 0; i < self->channels; i++) {
- gchar *name = g_strdup_printf ("src%d", i);
-
- GstCaps *srccaps;
-
- GstStructure *s;
-
- pad = gst_pad_new_from_static_template (&src_template, name);
- g_free (name);
-
- /* Set channel position if we know it */
- if (self->keep_positions) {
- GstAudioChannelPosition pos[1] = { GST_AUDIO_CHANNEL_POSITION_NONE };
-
- srccaps = gst_caps_copy (caps);
- s = gst_caps_get_structure (srccaps, 0);
- if (self->pos)
- gst_audio_set_channel_positions (s, &self->pos[i]);
- else
- gst_audio_set_channel_positions (s, pos);
- } else {
- srccaps = caps;
- }
-
- gst_pad_set_caps (pad, srccaps);
- gst_pad_use_fixed_caps (pad);
- gst_pad_set_query_function (pad,
- GST_DEBUG_FUNCPTR (gst_deinterleave_src_query));
- gst_pad_set_active (pad, TRUE);
- gst_element_add_pad (GST_ELEMENT (self), pad);
- self->srcpads = g_list_prepend (self->srcpads, gst_object_ref (pad));
-
- if (self->keep_positions)
- gst_caps_unref (srccaps);
- }
-
- gst_element_no_more_pads (GST_ELEMENT (self));
- self->srcpads = g_list_reverse (self->srcpads);
-}
-
-static void
-gst_deinterleave_set_pads_caps (GstDeinterleave * self, GstCaps * caps)
-{
- GList *l;
-
- GstStructure *s;
-
- gint i;
-
- for (l = self->srcpads, i = 0; l; l = l->next, i++) {
- GstPad *pad = GST_PAD (l->data);
-
- GstCaps *srccaps;
-
- /* Set channel position if we know it */
- if (self->keep_positions) {
- GstAudioChannelPosition pos[1] = { GST_AUDIO_CHANNEL_POSITION_NONE };
-
- srccaps = gst_caps_copy (caps);
- s = gst_caps_get_structure (srccaps, 0);
- if (self->pos)
- gst_audio_set_channel_positions (s, &self->pos[i]);
- else
- gst_audio_set_channel_positions (s, pos);
- } else {
- srccaps = caps;
- }
-
- gst_pad_set_caps (pad, srccaps);
-
- if (self->keep_positions)
- gst_caps_unref (srccaps);
- }
-}
-
-static void
-gst_deinterleave_remove_pads (GstDeinterleave * self)
-{
- GList *l;
-
- GST_INFO_OBJECT (self, "removing pads");
-
- for (l = self->srcpads; l; l = l->next) {
- GstPad *pad = GST_PAD (l->data);
-
- gst_element_remove_pad (GST_ELEMENT_CAST (self), pad);
- gst_object_unref (pad);
- }
- g_list_free (self->srcpads);
- self->srcpads = NULL;
-
- gst_pad_set_caps (self->sink, NULL);
- gst_caps_replace (&self->sinkcaps, NULL);
-}
-
-static gboolean
-gst_deinterleave_set_process_function (GstDeinterleave * self, GstCaps * caps)
-{
- GstStructure *s;
-
- s = gst_caps_get_structure (caps, 0);
- if (!gst_structure_get_int (s, "width", &self->width))
- return FALSE;
-
- switch (self->width) {
- case 8:
- self->func = (GstDeinterleaveFunc) deinterleave_8;
- break;
- case 16:
- self->func = (GstDeinterleaveFunc) deinterleave_16;
- break;
- case 24:
- self->func = (GstDeinterleaveFunc) deinterleave_24;
- break;
- case 32:
- self->func = (GstDeinterleaveFunc) deinterleave_32;
- break;
- case 64:
- self->func = (GstDeinterleaveFunc) deinterleave_64;
- break;
- default:
- return FALSE;
- }
- return TRUE;
-}
-
-static gboolean
-gst_deinterleave_sink_setcaps (GstPad * pad, GstCaps * caps)
-{
- GstDeinterleave *self;
-
- GstCaps *srccaps;
-
- GstStructure *s;
-
- self = GST_DEINTERLEAVE (gst_pad_get_parent (pad));
-
- GST_DEBUG_OBJECT (self, "got caps: %" GST_PTR_FORMAT, caps);
-
- if (self->sinkcaps && !gst_caps_is_equal (caps, self->sinkcaps)) {
- gint new_channels, i;
-
- GstAudioChannelPosition *pos;
-
- gboolean same_layout = TRUE;
-
- s = gst_caps_get_structure (caps, 0);
-
- /* We allow caps changes as long as the number of channels doesn't change
- * and the channel positions stay the same. _getcaps() should've cared
- * for this already but better be safe.
- */
- if (!gst_structure_get_int (s, "channels", &new_channels) ||
- new_channels != self->channels ||
- !gst_deinterleave_set_process_function (self, caps))
- goto cannot_change_caps;
-
- /* Now check the channel positions. If we had no channel positions
- * and get them or the other way around things have changed.
- * If we had channel positions and get different ones things have
- * changed too of course
- */
- pos = gst_audio_get_channel_positions (s);
- if ((pos && !self->pos) || (!pos && self->pos))
- goto cannot_change_caps;
-
- if (pos) {
- for (i = 0; i < self->channels; i++) {
- if (self->pos[i] != pos[i]) {
- same_layout = FALSE;
- break;
- }
- }
- g_free (pos);
- if (!same_layout)
- goto cannot_change_caps;
- }
-
- } else {
- s = gst_caps_get_structure (caps, 0);
-
- if (!gst_structure_get_int (s, "channels", &self->channels))
- goto no_channels;
-
- if (!gst_deinterleave_set_process_function (self, caps))
- goto unsupported_caps;
-
- self->pos = gst_audio_get_channel_positions (s);
- }
-
- gst_caps_replace (&self->sinkcaps, caps);
-
- /* Get srcpad caps */
- srccaps = gst_caps_copy (caps);
- s = gst_caps_get_structure (srccaps, 0);
- gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL);
- gst_structure_remove_field (s, "channel-positions");
-
- /* If we already have pads, update the caps otherwise
- * add new pads */
- if (self->srcpads) {
- gst_deinterleave_set_pads_caps (self, srccaps);
- } else {
- gst_deinterleave_add_new_pads (self, srccaps);
- }
-
- gst_caps_unref (srccaps);
- gst_object_unref (self);
-
- return TRUE;
-
-cannot_change_caps:
- {
- GST_ERROR_OBJECT (self, "can't set new caps: %" GST_PTR_FORMAT, caps);
- gst_object_unref (self);
- return FALSE;
- }
-unsupported_caps:
- {
- GST_ERROR_OBJECT (self, "caps not supported: %" GST_PTR_FORMAT, caps);
- gst_object_unref (self);
- return FALSE;
- }
-no_channels:
- {
- GST_ERROR_OBJECT (self, "invalid caps");
- gst_object_unref (self);
- return FALSE;
- }
-}
-
-static void
-__remove_channels (GstCaps * caps)
-{
- GstStructure *s;
-
- gint i, size;
-
- size = gst_caps_get_size (caps);
- for (i = 0; i < size; i++) {
- s = gst_caps_get_structure (caps, i);
- gst_structure_remove_field (s, "channel-positions");
- gst_structure_remove_field (s, "channels");
- }
-}
-
-static void
-__set_channels (GstCaps * caps, gint channels)
-{
- GstStructure *s;
-
- gint i, size;
-
- size = gst_caps_get_size (caps);
- for (i = 0; i < size; i++) {
- s = gst_caps_get_structure (caps, i);
- if (channels > 0)
- gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL);
- else
- gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
- }
-}
-
-static GstCaps *
-gst_deinterleave_sink_getcaps (GstPad * pad)
-{
- GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad));
-
- GstCaps *ret;
-
- GList *l;
-
- GST_OBJECT_LOCK (self);
- /* Intersect all of our pad template caps with the peer caps of the pad
- * to get all formats that are possible up- and downstream.
- *
- * For the pad for which the caps are requested we don't remove the channel
- * informations as they must be in the returned caps and incompatibilities
- * will be detected here already
- */
- ret = gst_caps_new_any ();
- for (l = GST_ELEMENT (self)->pads; l != NULL; l = l->next) {
- GstPad *ourpad = GST_PAD (l->data);
-
- GstCaps *peercaps = NULL, *ourcaps;
-
- ourcaps = gst_caps_copy (gst_pad_get_pad_template_caps (ourpad));
-
- if (pad == ourpad) {
- if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK)
- __set_channels (ourcaps, self->channels);
- else
- __set_channels (ourcaps, 1);
- } else {
- __remove_channels (ourcaps);
- /* Only ask for peer caps for other pads than pad
- * as otherwise gst_pad_peer_get_caps() might call
- * back into this function and deadlock
- */
- peercaps = gst_pad_peer_get_caps (ourpad);
- }
-
- /* If the peer exists and has caps add them to the intersection,
- * otherwise assume that the peer accepts everything */
- if (peercaps) {
- GstCaps *intersection;
-
- GstCaps *oldret = ret;
-
- __remove_channels (peercaps);
-
- intersection = gst_caps_intersect (peercaps, ourcaps);
-
- ret = gst_caps_intersect (ret, intersection);
- gst_caps_unref (intersection);
- gst_caps_unref (peercaps);
- gst_caps_unref (oldret);
- } else {
- GstCaps *oldret = ret;
-
- ret = gst_caps_intersect (ret, ourcaps);
- gst_caps_unref (oldret);
- }
- gst_caps_unref (ourcaps);
- }
- GST_OBJECT_UNLOCK (self);
-
- gst_object_unref (self);
-
- GST_DEBUG_OBJECT (pad, "Intersected caps to %" GST_PTR_FORMAT, ret);
-
- return ret;
-}
-
-static gboolean
-gst_deinterleave_sink_event (GstPad * pad, GstEvent * event)
-{
- GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad));
-
- gboolean ret;
-
- GST_DEBUG ("Got %s event on pad %s:%s", GST_EVENT_TYPE_NAME (event),
- GST_DEBUG_PAD_NAME (pad));
-
- /* Send FLUSH_STOP, FLUSH_START and EOS immediately, no matter if
- * we have src pads already or not. Queue all other events and
- * push them after we have src pads
- */
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_FLUSH_STOP:
- case GST_EVENT_FLUSH_START:
- case GST_EVENT_EOS:
- ret = gst_pad_event_default (pad, event);
- break;
- default:
- if (self->srcpads) {
- ret = gst_pad_event_default (pad, event);
- } else {
- GST_OBJECT_LOCK (self);
- self->pending_events = g_list_append (self->pending_events, event);
- GST_OBJECT_UNLOCK (self);
- ret = TRUE;
- }
- break;
- }
-
- gst_object_unref (self);
-
- return ret;
-}
-
-static gboolean
-gst_deinterleave_src_query (GstPad * pad, GstQuery * query)
-{
- GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad));
-
- gboolean res;
-
- res = gst_pad_query_default (pad, query);
-
- if (res && GST_QUERY_TYPE (query) == GST_QUERY_DURATION) {
- GstFormat format;
-
- gint64 dur;
-
- gst_query_parse_duration (query, &format, &dur);
-
- /* Need to divide by the number of channels in byte format
- * to get the correct value. All other formats should be fine
- */
- if (format == GST_FORMAT_BYTES && dur != -1)
- gst_query_set_duration (query, format, dur / self->channels);
- } else if (res && GST_QUERY_TYPE (query) == GST_QUERY_POSITION) {
- GstFormat format;
-
- gint64 pos;
-
- gst_query_parse_position (query, &format, &pos);
-
- /* Need to divide by the number of channels in byte format
- * to get the correct value. All other formats should be fine
- */
- if (format == GST_FORMAT_BYTES && pos != -1)
- gst_query_set_position (query, format, pos / self->channels);
- }
-
- gst_object_unref (self);
- return res;
-}
-
-static void
-gst_deinterleave_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
- GstDeinterleave *self = GST_DEINTERLEAVE (object);
-
- switch (prop_id) {
- case PROP_KEEP_POSITIONS:
- self->keep_positions = g_value_get_boolean (value);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static void
-gst_deinterleave_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec)
-{
- GstDeinterleave *self = GST_DEINTERLEAVE (object);
-
- switch (prop_id) {
- case PROP_KEEP_POSITIONS:
- g_value_set_boolean (value, self->keep_positions);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static GstFlowReturn
-gst_deinterleave_process (GstDeinterleave * self, GstBuffer * buf)
-{
- GstFlowReturn ret = GST_FLOW_OK;
-
- guint channels = self->channels;
-
- guint pads_pushed = 0, buffers_allocated = 0;
-
- guint nframes = GST_BUFFER_SIZE (buf) / channels / (self->width / 8);
-
- guint bufsize = nframes * (self->width / 8);
-
- guint i;
-
- GList *srcs;
-
- GstBuffer **buffers_out = g_new0 (GstBuffer *, channels);
-
- guint8 *in, *out;
-
- /* Send any pending events to all src pads */
- GST_OBJECT_LOCK (self);
- if (self->pending_events) {
- GList *events;
-
- GstEvent *event;
-
- GST_DEBUG_OBJECT (self, "Sending pending events to all src pads");
-
- for (events = self->pending_events; events != NULL; events = events->next) {
- event = GST_EVENT (events->data);
-
- for (srcs = self->srcpads; srcs != NULL; srcs = srcs->next)
- gst_pad_push_event (GST_PAD (srcs->data), gst_event_ref (event));
- gst_event_unref (event);
- }
-
- g_list_free (self->pending_events);
- self->pending_events = NULL;
- }
- GST_OBJECT_UNLOCK (self);
-
- /* Allocate buffers */
- for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) {
- GstPad *pad = (GstPad *) srcs->data;
-
- buffers_out[i] = NULL;
- ret =
- gst_pad_alloc_buffer (pad, GST_BUFFER_OFFSET_NONE, bufsize,
- GST_PAD_CAPS (pad), &buffers_out[i]);
-
- /* Make sure we got a correct buffer. The only other case we allow
- * here is an unliked pad */
- if (ret != GST_FLOW_OK && ret != GST_FLOW_NOT_LINKED)
- goto alloc_buffer_failed;
- else if (buffers_out[i] && GST_BUFFER_SIZE (buffers_out[i]) != bufsize)
- goto alloc_buffer_bad_size;
- else if (buffers_out[i] &&
- !gst_caps_is_equal (GST_BUFFER_CAPS (buffers_out[i]),
- GST_PAD_CAPS (pad)))
- goto invalid_caps;
-
- if (buffers_out[i]) {
- gst_buffer_copy_metadata (buffers_out[i], buf,
- GST_BUFFER_COPY_TIMESTAMPS | GST_BUFFER_COPY_FLAGS);
- buffers_allocated++;
- }
- }
-
- /* Return NOT_LINKED if no pad was linked */
- if (!buffers_allocated) {
- GST_WARNING_OBJECT (self,
- "Couldn't allocate any buffers because no pad was linked");
- ret = GST_FLOW_NOT_LINKED;
- goto done;
- }
-
- /* deinterleave */
- for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) {
- GstPad *pad = (GstPad *) srcs->data;
-
- in = (guint8 *) GST_BUFFER_DATA (buf);
- in += i * (self->width / 8);
- if (buffers_out[i]) {
- out = (guint8 *) GST_BUFFER_DATA (buffers_out[i]);
-
- self->func (out, in, channels, nframes);
-
- ret = gst_pad_push (pad, buffers_out[i]);
- buffers_out[i] = NULL;
- if (ret == GST_FLOW_OK)
- pads_pushed++;
- else if (ret == GST_FLOW_NOT_LINKED)
- ret = GST_FLOW_OK;
- else
- goto push_failed;
- }
- }
-
- /* Return NOT_LINKED if no pad was linked */
- if (!pads_pushed)
- ret = GST_FLOW_NOT_LINKED;
-
-done:
- gst_buffer_unref (buf);
- g_free (buffers_out);
- return ret;
-
-alloc_buffer_failed:
- {
- GST_WARNING ("gst_pad_alloc_buffer() returned %s", gst_flow_get_name (ret));
- goto clean_buffers;
-
- }
-alloc_buffer_bad_size:
- {
- GST_WARNING ("called alloc_buffer(), but didn't get requested bytes");
- ret = GST_FLOW_NOT_NEGOTIATED;
- goto clean_buffers;
- }
-invalid_caps:
- {
- GST_WARNING ("called alloc_buffer(), but didn't get requested caps");
- ret = GST_FLOW_NOT_NEGOTIATED;
- goto clean_buffers;
- }
-push_failed:
- {
- GST_DEBUG ("push() failed, flow = %s", gst_flow_get_name (ret));
- goto clean_buffers;
- }
-clean_buffers:
- {
- for (i = 0; i < channels; i++) {
- if (buffers_out[i])
- gst_buffer_unref (buffers_out[i]);
- }
- gst_buffer_unref (buf);
- g_free (buffers_out);
- return ret;
- }
-}
-
-static GstFlowReturn
-gst_deinterleave_chain (GstPad * pad, GstBuffer * buffer)
-{
- GstDeinterleave *self = GST_DEINTERLEAVE (GST_PAD_PARENT (pad));
-
- GstFlowReturn ret;
-
- g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED);
- g_return_val_if_fail (self->width > 0, GST_FLOW_NOT_NEGOTIATED);
- g_return_val_if_fail (self->channels > 0, GST_FLOW_NOT_NEGOTIATED);
-
- ret = gst_deinterleave_process (self, buffer);
-
- if (ret != GST_FLOW_OK)
- GST_DEBUG_OBJECT (self, "flow return: %s", gst_flow_get_name (ret));
-
- return ret;
-}
-
-static gboolean
-gst_deinterleave_sink_activate_push (GstPad * pad, gboolean active)
-{
- GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad));
-
- /* Reset everything when the pad is deactivated */
- if (!active) {
- gst_deinterleave_remove_pads (self);
- if (self->pos) {
- g_free (self->pos);
- self->pos = NULL;
- }
- self->channels = 0;
- self->width = 0;
- self->func = NULL;
-
- if (self->pending_events) {
- g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref,
- NULL);
- g_list_free (self->pending_events);
- self->pending_events = NULL;
- }
- }
-
- gst_object_unref (self);
-
- return TRUE;
-}
diff --git a/gst/interleave/deinterleave.h b/gst/interleave/deinterleave.h
deleted file mode 100644
index fe8ec75d..00000000
--- a/gst/interleave/deinterleave.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- * 2000 Wim Taymans <wtay@chello.be>
- * 2005 Wim Taymans <wim@fluendo.com>
- * 2007 Andy Wingo <wingo at pobox.com>
- * 2008 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * deinterleave.c: deinterleave samples
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 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
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifndef __DEINTERLEAVE_H__
-#define __DEINTERLEAVE_H__
-
-G_BEGIN_DECLS
-
-#include <gst/gst.h>
-#include <gst/audio/multichannel.h>
-
-#define GST_TYPE_DEINTERLEAVE (gst_deinterleave_get_type())
-#define GST_DEINTERLEAVE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_DEINTERLEAVE,GstDeinterleave))
-#define GST_DEINTERLEAVE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_DEINTERLEAVE,GstDeinterleaveClass))
-#define GST_DEINTERLEAVE_GET_CLASS(obj) \
- (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_DEINTERLEAVE,GstDeinterleaveClass))
-#define GST_IS_DEINTERLEAVE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_DEINTERLEAVE))
-#define GST_IS_DEINTERLEAVE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_DEINTERLEAVE))
-
-typedef struct _GstDeinterleave GstDeinterleave;
-typedef struct _GstDeinterleaveClass GstDeinterleaveClass;
-
-typedef void (*GstDeinterleaveFunc) (gpointer out, gpointer in, guint stride, guint nframes);
-
-struct _GstDeinterleave
-{
- GstElement element;
-
- /*< private > */
- GList *srcpads;
- GstCaps *sinkcaps;
- gint channels;
- GstAudioChannelPosition *pos;
- gboolean keep_positions;
-
- GstPad *sink;
-
- gint width;
- GstDeinterleaveFunc func;
-
- GList *pending_events;
-};
-
-struct _GstDeinterleaveClass
-{
- GstElementClass parent_class;
-};
-
-GType gst_deinterleave_get_type (void);
-
-G_END_DECLS
-
-#endif /* __DEINTERLEAVE_H__ */
diff --git a/gst/interleave/interleave.c b/gst/interleave/interleave.c
deleted file mode 100644
index 831e928f..00000000
--- a/gst/interleave/interleave.c
+++ /dev/null
@@ -1,1352 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- * 2000 Wim Taymans <wtay@chello.be>
- * 2005 Wim Taymans <wim@fluendo.com>
- * 2007 Andy Wingo <wingo at pobox.com>
- * 2008 Sebastian Dröge <slomo@circular-chaos.rg>
- *
- * interleave.c: interleave samples, mostly based on adder.
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 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
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-/* TODO:
- * - handle caps changes
- * - handle more queries/events
- */
-
-/**
- * SECTION:element-interleave
- * @see_also: deinterleave
- *
- * Merges separate mono inputs into one interleaved stream.
- *
- * This element handles all raw floating point sample formats and all signed integer sample formats. The first
- * caps on one of the sinkpads will set the caps of the output so usually an audioconvert element should be
- * placed before every sinkpad of interleave.
- *
- * It's possible to change the number of channels while the pipeline is running by adding or removing
- * some of the request pads but this will change the caps of the output buffers. Changing the input
- * caps is _not_ supported yet.
- *
- * The channel number of every sinkpad in the out can be retrieved from the "channel" property of the pad.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch-0.10 filesrc location=file.mp3 ! decodebin ! audioconvert ! "audio/x-raw-int,channels=2" ! deinterleave name=d interleave name=i ! audioconvert ! wavenc ! filesink location=test.wav d.src0 ! queue ! audioconvert ! i.sink1 d.src1 ! queue ! audioconvert ! i.sink0
- * ]| Decodes and deinterleaves a Stereo MP3 file into separate channels and
- * then interleaves the channels again to a WAV file with the channel with the
- * channels exchanged.
- * |[
- * gst-launch-0.10 interleave name=i ! audioconvert ! wavenc ! filesink location=file.wav filesrc location=file1.wav ! decodebin ! audioconvert ! "audio/x-raw-int,channels=1" ! queue ! i.sink0 filesrc location=file2.wav ! decodebin ! audioconvert ! "audio/x-raw-int,channels=1" ! queue ! i.sink1
- * ]| Interleaves two Mono WAV files to a single Stereo WAV file.
- * </refsect2>
- */
-
-#ifdef HAVE_CONFIG_H
-# include "config.h"
-#endif
-
-#include <gst/gst.h>
-#include <string.h>
-#include "interleave.h"
-
-#include <gst/audio/multichannel.h>
-
-GST_DEBUG_CATEGORY_STATIC (gst_interleave_debug);
-#define GST_CAT_DEFAULT gst_interleave_debug
-
-static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink%d",
- GST_PAD_SINK,
- GST_PAD_REQUEST,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) 1, "
- "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
- "width = (int) { 8, 16, 24, 32 }, "
- "depth = (int) [ 1, 32 ], "
- "signed = (boolean) true; "
- "audio/x-raw-float, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) 1, "
- "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, "
- "width = (int) { 32, 64 }")
- );
-
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-int, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, MAX ], "
- "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, "
- "width = (int) { 8, 16, 24, 32 }, "
- "depth = (int) [ 1, 32 ], "
- "signed = (boolean) true; "
- "audio/x-raw-float, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, MAX ], "
- "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, "
- "width = (int) { 32, 64 }")
- );
-
-#define MAKE_FUNC(type) \
-static void interleave_##type (guint##type *out, guint##type *in, \
- guint stride, guint nframes) \
-{ \
- gint i; \
- \
- for (i = 0; i < nframes; i++) { \
- *out = in[i]; \
- out += stride; \
- } \
-}
-
-MAKE_FUNC (8);
-MAKE_FUNC (16);
-MAKE_FUNC (32);
-MAKE_FUNC (64);
-
-static void
-interleave_24 (guint8 * out, guint8 * in, guint stride, guint nframes)
-{
- gint i;
-
- for (i = 0; i < nframes; i++) {
- memcpy (out, in, 3);
- out += stride * 3;
- in += 3;
- }
-}
-
-typedef struct
-{
- GstPad parent;
- guint channel;
-} GstInterleavePad;
-
-enum
-{
- PROP_PAD_0,
- PROP_PAD_CHANNEL
-};
-
-static void gst_interleave_pad_class_init (GstPadClass * klass);
-
-#define GST_TYPE_INTERLEAVE_PAD (gst_interleave_pad_get_type())
-#define GST_INTERLEAVE_PAD(pad) (G_TYPE_CHECK_INSTANCE_CAST((pad),GST_TYPE_INTERLEAVE_PAD,GstInterleavePad))
-#define GST_INTERLEAVE_PAD_CAST(pad) ((GstInterleavePad *) pad)
-#define GST_IS_INTERLEAVE_PAD(pad) (G_TYPE_CHECK_INSTANCE_TYPE((pad),GST_TYPE_INTERLEAVE_PAD))
-static GType
-gst_interleave_pad_get_type (void)
-{
- static GType type = 0;
-
- if (G_UNLIKELY (type == 0)) {
- type = g_type_register_static_simple (GST_TYPE_PAD,
- g_intern_static_string ("GstInterleavePad"), sizeof (GstPadClass),
- (GClassInitFunc) gst_interleave_pad_class_init,
- sizeof (GstInterleavePad), NULL, 0);
- }
- return type;
-}
-
-static void
-gst_interleave_pad_get_property (GObject * object,
- guint prop_id, GValue * value, GParamSpec * pspec)
-{
- GstInterleavePad *self = GST_INTERLEAVE_PAD (object);
-
- switch (prop_id) {
- case PROP_PAD_CHANNEL:
- g_value_set_uint (value, self->channel);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static void
-gst_interleave_pad_class_init (GstPadClass * klass)
-{
- GObjectClass *gobject_class = (GObjectClass *) klass;
-
- gobject_class->get_property = gst_interleave_pad_get_property;
-
- g_object_class_install_property (gobject_class,
- PROP_PAD_CHANNEL,
- g_param_spec_uint ("channel",
- "Channel number",
- "Number of the channel of this pad in the output", 0, G_MAXUINT, 0,
- G_PARAM_READABLE | G_PARAM_STATIC_STRINGS));
-}
-
-GST_BOILERPLATE (GstInterleave, gst_interleave, GstElement, GST_TYPE_ELEMENT);
-
-enum
-{
- PROP_0,
- PROP_CHANNEL_POSITIONS,
- PROP_CHANNEL_POSITIONS_FROM_INPUT
-};
-
-static void gst_interleave_set_property (GObject * object,
- guint prop_id, const GValue * value, GParamSpec * pspec);
-static void gst_interleave_get_property (GObject * object,
- guint prop_id, GValue * value, GParamSpec * pspec);
-
-static GstPad *gst_interleave_request_new_pad (GstElement * element,
- GstPadTemplate * templ, const gchar * name);
-static void gst_interleave_release_pad (GstElement * element, GstPad * pad);
-
-static GstStateChangeReturn gst_interleave_change_state (GstElement * element,
- GstStateChange transition);
-
-static gboolean gst_interleave_src_query (GstPad * pad, GstQuery * query);
-
-static gboolean gst_interleave_src_event (GstPad * pad, GstEvent * event);
-
-static gboolean gst_interleave_sink_event (GstPad * pad, GstEvent * event);
-
-static gboolean gst_interleave_sink_setcaps (GstPad * pad, GstCaps * caps);
-
-static GstCaps *gst_interleave_sink_getcaps (GstPad * pad);
-
-static GstFlowReturn gst_interleave_collected (GstCollectPads * pads,
- GstInterleave * self);
-
-static void
-gst_interleave_finalize (GObject * object)
-{
- GstInterleave *self = GST_INTERLEAVE (object);
-
- if (self->collect) {
- gst_object_unref (self->collect);
- self->collect = NULL;
- }
-
- if (self->channel_positions
- && self->channel_positions != self->input_channel_positions) {
- g_value_array_free (self->channel_positions);
- self->channel_positions = NULL;
- }
-
- if (self->input_channel_positions) {
- g_value_array_free (self->input_channel_positions);
- self->input_channel_positions = NULL;
- }
-
- gst_caps_replace (&self->sinkcaps, NULL);
-
- G_OBJECT_CLASS (parent_class)->finalize (object);
-}
-
-static gboolean
-gst_interleave_check_channel_positions (GValueArray * positions)
-{
- gint i;
-
- guint channels;
-
- GstAudioChannelPosition *pos;
-
- gboolean ret;
-
- channels = positions->n_values;
- pos = g_new (GstAudioChannelPosition, positions->n_values);
-
- for (i = 0; i < channels; i++) {
- GValue *v = g_value_array_get_nth (positions, i);
-
- pos[i] = g_value_get_enum (v);
- }
-
- ret = gst_audio_check_channel_positions (pos, channels);
- g_free (pos);
-
- return ret;
-}
-
-static void
-gst_interleave_set_channel_positions (GstInterleave * self, GstStructure * s)
-{
- GValue pos_array = { 0, };
- gint i;
-
- g_value_init (&pos_array, GST_TYPE_ARRAY);
-
- if (self->channel_positions
- && self->channels == self->channel_positions->n_values
- && gst_interleave_check_channel_positions (self->channel_positions)) {
- GST_DEBUG_OBJECT (self, "Using provided channel positions");
- for (i = 0; i < self->channels; i++)
- gst_value_array_append_value (&pos_array,
- g_value_array_get_nth (self->channel_positions, i));
- } else {
- GValue pos_none = { 0, };
-
- GST_WARNING_OBJECT (self, "Using NONE channel positions");
-
- g_value_init (&pos_none, GST_TYPE_AUDIO_CHANNEL_POSITION);
- g_value_set_enum (&pos_none, GST_AUDIO_CHANNEL_POSITION_NONE);
-
- for (i = 0; i < self->channels; i++)
- gst_value_array_append_value (&pos_array, &pos_none);
-
- g_value_unset (&pos_none);
- }
- gst_structure_set_value (s, "channel-positions", &pos_array);
- g_value_unset (&pos_array);
-}
-
-static void
-gst_interleave_base_init (gpointer g_class)
-{
- gst_element_class_set_details_simple (g_class, "Audio interleaver",
- "Filter/Converter/Audio",
- "Folds many mono channels into one interleaved audio stream",
- "Andy Wingo <wingo at pobox.com>, "
- "Sebastian Dröge <slomo@circular-chaos.org>");
-
- gst_element_class_add_pad_template (g_class,
- gst_static_pad_template_get (&sink_template));
- gst_element_class_add_pad_template (g_class,
- gst_static_pad_template_get (&src_template));
-}
-
-static void
-gst_interleave_class_init (GstInterleaveClass * klass)
-{
- GstElementClass *gstelement_class;
-
- GObjectClass *gobject_class;
-
- gobject_class = G_OBJECT_CLASS (klass);
- gstelement_class = GST_ELEMENT_CLASS (klass);
-
- GST_DEBUG_CATEGORY_INIT (gst_interleave_debug, "interleave", 0,
- "interleave element");
-
- /* Reference GstInterleavePad class to have the type registered from
- * a threadsafe context
- */
- g_type_class_ref (GST_TYPE_INTERLEAVE_PAD);
-
- gobject_class->finalize = gst_interleave_finalize;
- gobject_class->set_property = gst_interleave_set_property;
- gobject_class->get_property = gst_interleave_get_property;
-
- /**
- * GstInterleave:channel-positions
- *
- * Channel positions: This property controls the channel positions
- * that are used on the src caps. The number of elements should be
- * the same as the number of sink pads and the array should contain
- * a valid list of channel positions. The n-th element of the array
- * is the position of the n-th sink pad.
- *
- * These channel positions will only be used if they're valid and the
- * number of elements is the same as the number of channels. If this
- * is not given a NONE layout will be used.
- *
- */
- g_object_class_install_property (gobject_class, PROP_CHANNEL_POSITIONS,
- g_param_spec_value_array ("channel-positions", "Channel positions",
- "Channel positions used on the output",
- g_param_spec_enum ("channel-position", "Channel position",
- "Channel position of the n-th input",
- GST_TYPE_AUDIO_CHANNEL_POSITION,
- GST_AUDIO_CHANNEL_POSITION_NONE,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS),
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
- /**
- * GstInterleave:channel-positions-from-input
- *
- * Channel positions from input: If this property is set to %TRUE the channel
- * positions will be taken from the input caps if valid channel positions for
- * the output can be constructed from them. If this is set to %TRUE setting the
- * channel-positions property overwrites this property again.
- *
- */
- g_object_class_install_property (gobject_class,
- PROP_CHANNEL_POSITIONS_FROM_INPUT,
- g_param_spec_boolean ("channel-positions-from-input",
- "Channel positions from input",
- "Take channel positions from the input", TRUE,
- G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
-
- gstelement_class->request_new_pad =
- GST_DEBUG_FUNCPTR (gst_interleave_request_new_pad);
- gstelement_class->release_pad =
- GST_DEBUG_FUNCPTR (gst_interleave_release_pad);
- gstelement_class->change_state =
- GST_DEBUG_FUNCPTR (gst_interleave_change_state);
-}
-
-static void
-gst_interleave_init (GstInterleave * self, GstInterleaveClass * klass)
-{
- self->src = gst_pad_new_from_static_template (&src_template, "src");
-
- gst_pad_set_query_function (self->src,
- GST_DEBUG_FUNCPTR (gst_interleave_src_query));
- gst_pad_set_event_function (self->src,
- GST_DEBUG_FUNCPTR (gst_interleave_src_event));
-
- gst_element_add_pad (GST_ELEMENT (self), self->src);
-
- self->collect = gst_collect_pads_new ();
- gst_collect_pads_set_function (self->collect,
- (GstCollectPadsFunction) gst_interleave_collected, self);
-
- self->input_channel_positions = g_value_array_new (0);
- self->channel_positions_from_input = TRUE;
- self->channel_positions = self->input_channel_positions;
-}
-
-static void
-gst_interleave_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
- GstInterleave *self = GST_INTERLEAVE (object);
-
- switch (prop_id) {
- case PROP_CHANNEL_POSITIONS:
- if (self->channel_positions &&
- self->channel_positions != self->input_channel_positions)
- g_value_array_free (self->channel_positions);
-
- self->channel_positions = g_value_dup_boxed (value);
- self->channel_positions_from_input = FALSE;
- break;
- case PROP_CHANNEL_POSITIONS_FROM_INPUT:
- self->channel_positions_from_input = g_value_get_boolean (value);
-
- if (self->channel_positions_from_input) {
- if (self->channel_positions &&
- self->channel_positions != self->input_channel_positions)
- g_value_array_free (self->channel_positions);
- self->channel_positions = self->input_channel_positions;
- }
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static void
-gst_interleave_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec)
-{
- GstInterleave *self = GST_INTERLEAVE (object);
-
- switch (prop_id) {
- case PROP_CHANNEL_POSITIONS:
- g_value_set_boxed (value, self->channel_positions);
- break;
- case PROP_CHANNEL_POSITIONS_FROM_INPUT:
- g_value_set_boolean (value, self->channel_positions_from_input);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static GstPad *
-gst_interleave_request_new_pad (GstElement * element, GstPadTemplate * templ,
- const gchar * req_name)
-{
- GstInterleave *self = GST_INTERLEAVE (element);
-
- GstPad *new_pad;
-
- gchar *pad_name;
-
- gint channels, padnumber;
- GValue val = { 0, };
-
- if (templ->direction != GST_PAD_SINK)
- goto not_sink_pad;
-
- channels = g_atomic_int_exchange_and_add (&self->channels, 1);
- padnumber = g_atomic_int_exchange_and_add (&self->padcounter, 1);
-
- pad_name = g_strdup_printf ("sink%d", padnumber);
- new_pad = GST_PAD_CAST (g_object_new (GST_TYPE_INTERLEAVE_PAD,
- "name", pad_name, "direction", templ->direction,
- "template", templ, NULL));
- GST_INTERLEAVE_PAD_CAST (new_pad)->channel = channels;
- GST_DEBUG_OBJECT (self, "requested new pad %s", pad_name);
- g_free (pad_name);
-
- gst_pad_set_setcaps_function (new_pad,
- GST_DEBUG_FUNCPTR (gst_interleave_sink_setcaps));
- gst_pad_set_getcaps_function (new_pad,
- GST_DEBUG_FUNCPTR (gst_interleave_sink_getcaps));
-
- gst_collect_pads_add_pad (self->collect, new_pad, sizeof (GstCollectData));
-
- /* FIXME: hacked way to override/extend the event function of
- * GstCollectPads; because it sets its own event function giving the
- * element no access to events */
- self->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (new_pad);
- gst_pad_set_event_function (new_pad,
- GST_DEBUG_FUNCPTR (gst_interleave_sink_event));
-
- if (!gst_element_add_pad (element, new_pad))
- goto could_not_add;
-
- g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION);
- g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_NONE);
- self->input_channel_positions =
- g_value_array_append (self->input_channel_positions, &val);
- g_value_unset (&val);
-
- /* Update the src caps if we already have them */
- if (self->sinkcaps) {
- GstCaps *srccaps;
-
- GstStructure *s;
-
- /* Take lock to make sure processing finishes first */
- GST_OBJECT_LOCK (self->collect);
-
- srccaps = gst_caps_copy (self->sinkcaps);
- s = gst_caps_get_structure (srccaps, 0);
-
- gst_structure_set (s, "channels", G_TYPE_INT, self->channels, NULL);
- gst_interleave_set_channel_positions (self, s);
-
- gst_pad_set_caps (self->src, srccaps);
- gst_caps_unref (srccaps);
-
- GST_OBJECT_UNLOCK (self->collect);
- }
-
- return new_pad;
-
- /* errors */
-not_sink_pad:
- {
- g_warning ("interleave: requested new pad that is not a SINK pad\n");
- return NULL;
- }
-could_not_add:
- {
- GST_DEBUG_OBJECT (self, "could not add pad %s", GST_PAD_NAME (new_pad));
- gst_collect_pads_remove_pad (self->collect, new_pad);
- gst_object_unref (new_pad);
- return NULL;
- }
-}
-
-static void
-gst_interleave_release_pad (GstElement * element, GstPad * pad)
-{
- GstInterleave *self = GST_INTERLEAVE (element);
-
- GList *l;
-
- g_return_if_fail (GST_IS_INTERLEAVE_PAD (pad));
-
- /* Take lock to make sure we're not changing this when processing buffers */
- GST_OBJECT_LOCK (self->collect);
-
- g_atomic_int_add (&self->channels, -1);
-
- g_value_array_remove (self->input_channel_positions,
- GST_INTERLEAVE_PAD_CAST (pad)->channel);
-
- /* Update channel numbers */
- GST_OBJECT_LOCK (self);
- for (l = GST_ELEMENT_CAST (self)->sinkpads; l != NULL; l = l->next) {
- GstInterleavePad *ipad = GST_INTERLEAVE_PAD (l->data);
-
- if (GST_INTERLEAVE_PAD_CAST (pad)->channel < ipad->channel)
- ipad->channel--;
- }
- GST_OBJECT_UNLOCK (self);
-
- /* Update the src caps if we already have them */
- if (self->sinkcaps) {
- if (self->channels > 0) {
- GstCaps *srccaps;
-
- GstStructure *s;
-
- srccaps = gst_caps_copy (self->sinkcaps);
- s = gst_caps_get_structure (srccaps, 0);
-
- gst_structure_set (s, "channels", G_TYPE_INT, self->channels, NULL);
- gst_interleave_set_channel_positions (self, s);
-
- gst_pad_set_caps (self->src, srccaps);
- gst_caps_unref (srccaps);
- } else {
- gst_caps_replace (&self->sinkcaps, NULL);
- gst_pad_set_caps (self->src, NULL);
- }
- }
-
- GST_OBJECT_UNLOCK (self->collect);
-
- gst_collect_pads_remove_pad (self->collect, pad);
- gst_element_remove_pad (element, pad);
-}
-
-static GstStateChangeReturn
-gst_interleave_change_state (GstElement * element, GstStateChange transition)
-{
- GstInterleave *self;
-
- GstStateChangeReturn ret;
-
- self = GST_INTERLEAVE (element);
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
- break;
- case GST_STATE_CHANGE_READY_TO_PAUSED:
- self->timestamp = 0;
- self->offset = 0;
- self->segment_pending = TRUE;
- self->segment_position = 0;
- self->segment_rate = 1.0;
- gst_segment_init (&self->segment, GST_FORMAT_UNDEFINED);
- gst_collect_pads_start (self->collect);
- break;
- case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
- break;
- default:
- break;
- }
-
- /* Stop before calling the parent's state change function as
- * GstCollectPads might take locks and we would deadlock in that
- * case
- */
- if (transition == GST_STATE_CHANGE_PAUSED_TO_READY)
- gst_collect_pads_stop (self->collect);
-
- ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
- switch (transition) {
- case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
- break;
- case GST_STATE_CHANGE_PAUSED_TO_READY:
- gst_pad_set_caps (self->src, NULL);
- gst_caps_replace (&self->sinkcaps, NULL);
- break;
- case GST_STATE_CHANGE_READY_TO_NULL:
- break;
- default:
- break;
- }
-
- return ret;
-}
-
-static void
-__remove_channels (GstCaps * caps)
-{
- GstStructure *s;
-
- gint i, size;
-
- size = gst_caps_get_size (caps);
- for (i = 0; i < size; i++) {
- s = gst_caps_get_structure (caps, i);
- gst_structure_remove_field (s, "channel-positions");
- gst_structure_remove_field (s, "channels");
- }
-}
-
-static void
-__set_channels (GstCaps * caps, gint channels)
-{
- GstStructure *s;
-
- gint i, size;
-
- size = gst_caps_get_size (caps);
- for (i = 0; i < size; i++) {
- s = gst_caps_get_structure (caps, i);
- if (channels > 0)
- gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL);
- else
- gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL);
- }
-}
-
-/* we can only accept caps that we and downstream can handle. */
-static GstCaps *
-gst_interleave_sink_getcaps (GstPad * pad)
-{
- GstInterleave *self = GST_INTERLEAVE (gst_pad_get_parent (pad));
-
- GstCaps *result, *peercaps, *sinkcaps;
-
- GST_OBJECT_LOCK (self);
-
- /* If we already have caps on one of the sink pads return them */
- if (self->sinkcaps) {
- result = gst_caps_copy (self->sinkcaps);
- } else {
- /* get the downstream possible caps */
- peercaps = gst_pad_peer_get_caps (self->src);
- /* get the allowed caps on this sinkpad */
- sinkcaps = gst_caps_copy (gst_pad_get_pad_template_caps (pad));
- __remove_channels (sinkcaps);
- if (peercaps) {
- __remove_channels (peercaps);
- /* if the peer has caps, intersect */
- GST_DEBUG_OBJECT (pad, "intersecting peer and template caps");
- result = gst_caps_intersect (peercaps, sinkcaps);
- gst_caps_unref (peercaps);
- gst_caps_unref (sinkcaps);
- } else {
- /* the peer has no caps (or there is no peer), just use the allowed caps
- * of this sinkpad. */
- GST_DEBUG_OBJECT (pad, "no peer caps, using sinkcaps");
- result = sinkcaps;
- }
- __set_channels (result, 1);
- }
-
- GST_OBJECT_UNLOCK (self);
-
- gst_object_unref (self);
-
- GST_DEBUG_OBJECT (pad, "Returning caps %" GST_PTR_FORMAT, result);
-
- return result;
-}
-
-static void
-gst_interleave_set_process_function (GstInterleave * self)
-{
- switch (self->width) {
- case 8:
- self->func = (GstInterleaveFunc) interleave_8;
- break;
- case 16:
- self->func = (GstInterleaveFunc) interleave_16;
- break;
- case 24:
- self->func = (GstInterleaveFunc) interleave_24;
- break;
- case 32:
- self->func = (GstInterleaveFunc) interleave_32;
- break;
- case 64:
- self->func = (GstInterleaveFunc) interleave_64;
- break;
- default:
- g_assert_not_reached ();
- break;
- }
-}
-
-static gboolean
-gst_interleave_sink_setcaps (GstPad * pad, GstCaps * caps)
-{
- GstInterleave *self;
-
- g_return_val_if_fail (GST_IS_INTERLEAVE_PAD (pad), FALSE);
-
- self = GST_INTERLEAVE (gst_pad_get_parent (pad));
-
- /* First caps that are set on a sink pad are used as output caps */
- /* TODO: handle caps changes */
- if (self->sinkcaps && !gst_caps_is_subset (caps, self->sinkcaps)) {
- goto cannot_change_caps;
- } else {
- GstCaps *srccaps;
-
- GstStructure *s;
-
- gboolean res;
-
- s = gst_caps_get_structure (caps, 0);
-
- if (!gst_structure_get_int (s, "width", &self->width))
- goto no_width;
-
- if (!gst_structure_get_int (s, "rate", &self->rate))
- goto no_rate;
-
- gst_interleave_set_process_function (self);
-
- if (gst_structure_has_field (s, "channel-positions")) {
- const GValue *pos_array;
-
- pos_array = gst_structure_get_value (s, "channel-positions");
- if (GST_VALUE_HOLDS_ARRAY (pos_array)
- && gst_value_array_get_size (pos_array) == 1) {
- const GValue *pos = gst_value_array_get_value (pos_array, 0);
-
- GValue *apos = g_value_array_get_nth (self->input_channel_positions,
- GST_INTERLEAVE_PAD_CAST (pad)->channel);
-
- g_value_set_enum (apos, g_value_get_enum (pos));
- }
- }
-
- srccaps = gst_caps_copy (caps);
- s = gst_caps_get_structure (srccaps, 0);
-
- gst_structure_set (s, "channels", G_TYPE_INT, self->channels, NULL);
- gst_interleave_set_channel_positions (self, s);
-
- res = gst_pad_set_caps (self->src, srccaps);
- gst_caps_unref (srccaps);
-
- if (!res)
- goto src_did_not_accept;
- }
-
- if (!self->sinkcaps) {
- GstCaps *sinkcaps = gst_caps_copy (caps);
-
- GstStructure *s = gst_caps_get_structure (sinkcaps, 0);
-
- gst_structure_remove_field (s, "channel-positions");
-
- gst_caps_replace (&self->sinkcaps, sinkcaps);
-
- gst_caps_unref (sinkcaps);
- }
-
- gst_object_unref (self);
-
- return TRUE;
-
-cannot_change_caps:
- {
- GST_DEBUG_OBJECT (self, "caps of %" GST_PTR_FORMAT " already set, can't "
- "change", self->sinkcaps);
- gst_object_unref (self);
- return FALSE;
- }
-src_did_not_accept:
- {
- GST_DEBUG_OBJECT (self, "src did not accept setcaps()");
- gst_object_unref (self);
- return FALSE;
- }
-no_width:
- {
- GST_WARNING_OBJECT (self, "caps did not have width: %" GST_PTR_FORMAT,
- caps);
- gst_object_unref (self);
- return FALSE;
- }
-no_rate:
- {
- GST_WARNING_OBJECT (self, "caps did not have rate: %" GST_PTR_FORMAT, caps);
- gst_object_unref (self);
- return FALSE;
- }
-}
-
-static gboolean
-gst_interleave_sink_event (GstPad * pad, GstEvent * event)
-{
- GstInterleave *self = GST_INTERLEAVE (gst_pad_get_parent (pad));
-
- gboolean ret;
-
- GST_DEBUG ("Got %s event on pad %s:%s", GST_EVENT_TYPE_NAME (event),
- GST_DEBUG_PAD_NAME (pad));
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_FLUSH_STOP:
- /* mark a pending new segment. This event is synchronized
- * with the streaming thread so we can safely update the
- * variable without races. It's somewhat weird because we
- * assume the collectpads forwarded the FLUSH_STOP past us
- * and downstream (using our source pad, the bastard!).
- */
- self->segment_pending = TRUE;
- break;
- default:
- break;
- }
-
- /* now GstCollectPads can take care of the rest, e.g. EOS */
- ret = self->collect_event (pad, event);
-
- gst_object_unref (self);
- return ret;
-}
-
-static gboolean
-gst_interleave_src_query_duration (GstInterleave * self, GstQuery * query)
-{
- gint64 max;
-
- gboolean res;
-
- GstFormat format;
-
- GstIterator *it;
-
- gboolean done;
-
- /* parse format */
- gst_query_parse_duration (query, &format, NULL);
-
- max = -1;
- res = TRUE;
- done = FALSE;
-
- /* Take maximum of all durations */
- it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (self));
- while (!done) {
- GstIteratorResult ires;
-
- gpointer item;
-
- ires = gst_iterator_next (it, &item);
- switch (ires) {
- case GST_ITERATOR_DONE:
- done = TRUE;
- break;
- case GST_ITERATOR_OK:
- {
- GstPad *pad = GST_PAD_CAST (item);
-
- gint64 duration;
-
- /* ask sink peer for duration */
- res &= gst_pad_query_peer_duration (pad, &format, &duration);
- /* take max from all valid return values */
- if (res) {
- /* valid unknown length, stop searching */
- if (duration == -1) {
- max = duration;
- done = TRUE;
- }
- /* else see if bigger than current max */
- else if (duration > max)
- max = duration;
- }
- gst_object_unref (pad);
- break;
- }
- case GST_ITERATOR_RESYNC:
- max = -1;
- res = TRUE;
- gst_iterator_resync (it);
- break;
- default:
- res = FALSE;
- done = TRUE;
- break;
- }
- }
- gst_iterator_free (it);
-
- if (res) {
- /* If in bytes format we have to multiply with the number of channels
- * to get the correct results. All other formats should be fine */
- if (format == GST_FORMAT_BYTES && max != -1)
- max *= self->channels;
-
- /* and store the max */
- GST_DEBUG_OBJECT (self, "Total duration in format %s: %"
- GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
- gst_query_set_duration (query, format, max);
- }
-
- return res;
-}
-
-static gboolean
-gst_interleave_src_query_latency (GstInterleave * self, GstQuery * query)
-{
- GstClockTime min, max;
-
- gboolean live;
-
- gboolean res;
-
- GstIterator *it;
-
- gboolean done;
-
- res = TRUE;
- done = FALSE;
-
- live = FALSE;
- min = 0;
- max = GST_CLOCK_TIME_NONE;
-
- /* Take maximum of all latency values */
- it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (self));
- while (!done) {
- GstIteratorResult ires;
-
- gpointer item;
-
- ires = gst_iterator_next (it, &item);
- switch (ires) {
- case GST_ITERATOR_DONE:
- done = TRUE;
- break;
- case GST_ITERATOR_OK:
- {
- GstPad *pad = GST_PAD_CAST (item);
-
- GstQuery *peerquery;
-
- GstClockTime min_cur, max_cur;
-
- gboolean live_cur;
-
- peerquery = gst_query_new_latency ();
-
- /* Ask peer for latency */
- res &= gst_pad_peer_query (pad, peerquery);
-
- /* take max from all valid return values */
- if (res) {
- gst_query_parse_latency (peerquery, &live_cur, &min_cur, &max_cur);
-
- if (min_cur > min)
- min = min_cur;
-
- if (max_cur != GST_CLOCK_TIME_NONE &&
- ((max != GST_CLOCK_TIME_NONE && max_cur > max) ||
- (max == GST_CLOCK_TIME_NONE)))
- max = max_cur;
-
- live = live || live_cur;
- }
-
- gst_query_unref (peerquery);
- gst_object_unref (pad);
- break;
- }
- case GST_ITERATOR_RESYNC:
- live = FALSE;
- min = 0;
- max = GST_CLOCK_TIME_NONE;
- res = TRUE;
- gst_iterator_resync (it);
- break;
- default:
- res = FALSE;
- done = TRUE;
- break;
- }
- }
- gst_iterator_free (it);
-
- if (res) {
- /* store the results */
- GST_DEBUG_OBJECT (self, "Calculated total latency: live %s, min %"
- GST_TIME_FORMAT ", max %" GST_TIME_FORMAT,
- (live ? "yes" : "no"), GST_TIME_ARGS (min), GST_TIME_ARGS (max));
- gst_query_set_latency (query, live, min, max);
- }
-
- return res;
-}
-
-static gboolean
-gst_interleave_src_query (GstPad * pad, GstQuery * query)
-{
- GstInterleave *self = GST_INTERLEAVE (gst_pad_get_parent (pad));
-
- gboolean res = FALSE;
-
- switch (GST_QUERY_TYPE (query)) {
- case GST_QUERY_POSITION:
- {
- GstFormat format;
-
- gst_query_parse_position (query, &format, NULL);
-
- switch (format) {
- case GST_FORMAT_TIME:
- /* FIXME, bring to stream time, might be tricky */
- gst_query_set_position (query, format, self->timestamp);
- res = TRUE;
- break;
- case GST_FORMAT_BYTES:
- gst_query_set_position (query, format,
- self->offset * self->channels * self->width);
- res = TRUE;
- break;
- case GST_FORMAT_DEFAULT:
- gst_query_set_position (query, format, self->offset);
- res = TRUE;
- break;
- default:
- break;
- }
- break;
- }
- case GST_QUERY_DURATION:
- res = gst_interleave_src_query_duration (self, query);
- break;
- case GST_QUERY_LATENCY:
- res = gst_interleave_src_query_latency (self, query);
- break;
- default:
- /* FIXME, needs a custom query handler because we have multiple
- * sinkpads */
- res = gst_pad_query_default (pad, query);
- break;
- }
-
- gst_object_unref (self);
- return res;
-}
-
-static gboolean
-forward_event_func (GstPad * pad, GValue * ret, GstEvent * event)
-{
- gst_event_ref (event);
- GST_LOG_OBJECT (pad, "About to send event %s", GST_EVENT_TYPE_NAME (event));
- if (!gst_pad_push_event (pad, event)) {
- g_value_set_boolean (ret, FALSE);
- GST_WARNING_OBJECT (pad, "Sending event %p (%s) failed.",
- event, GST_EVENT_TYPE_NAME (event));
- } else {
- GST_LOG_OBJECT (pad, "Sent event %p (%s).",
- event, GST_EVENT_TYPE_NAME (event));
- }
- gst_object_unref (pad);
- return TRUE;
-}
-
-static gboolean
-forward_event (GstInterleave * self, GstEvent * event)
-{
- gboolean ret;
-
- GstIterator *it;
- GValue vret = { 0 };
-
- GST_LOG_OBJECT (self, "Forwarding event %p (%s)", event,
- GST_EVENT_TYPE_NAME (event));
-
- ret = TRUE;
-
- g_value_init (&vret, G_TYPE_BOOLEAN);
- g_value_set_boolean (&vret, TRUE);
- it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (self));
- gst_iterator_fold (it, (GstIteratorFoldFunction) forward_event_func, &vret,
- event);
- gst_iterator_free (it);
- gst_event_unref (event);
-
- ret = g_value_get_boolean (&vret);
-
- return ret;
-}
-
-
-static gboolean
-gst_interleave_src_event (GstPad * pad, GstEvent * event)
-{
- GstInterleave *self = GST_INTERLEAVE (gst_pad_get_parent (pad));
-
- gboolean result;
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_QOS:
- /* QoS might be tricky */
- result = FALSE;
- break;
- case GST_EVENT_SEEK:
- {
- GstSeekFlags flags;
-
- GstSeekType curtype;
-
- gint64 cur;
-
- /* parse the seek parameters */
- gst_event_parse_seek (event, &self->segment_rate, NULL, &flags, &curtype,
- &cur, NULL, NULL);
-
- /* check if we are flushing */
- if (flags & GST_SEEK_FLAG_FLUSH) {
- /* make sure we accept nothing anymore and return WRONG_STATE */
- gst_collect_pads_set_flushing (self->collect, TRUE);
-
- /* flushing seek, start flush downstream, the flush will be done
- * when all pads received a FLUSH_STOP. */
- gst_pad_push_event (self->src, gst_event_new_flush_start ());
- }
-
- /* now wait for the collected to be finished and mark a new
- * segment */
- GST_OBJECT_LOCK (self->collect);
- if (curtype == GST_SEEK_TYPE_SET)
- self->segment_position = cur;
- else
- self->segment_position = 0;
- self->segment_pending = TRUE;
- GST_OBJECT_UNLOCK (self->collect);
-
- result = forward_event (self, event);
- break;
- }
- case GST_EVENT_NAVIGATION:
- /* navigation is rather pointless. */
- result = FALSE;
- break;
- default:
- /* just forward the rest for now */
- result = forward_event (self, event);
- break;
- }
- gst_object_unref (self);
-
- return result;
-}
-
-static GstFlowReturn
-gst_interleave_collected (GstCollectPads * pads, GstInterleave * self)
-{
- guint size;
-
- GstBuffer *outbuf;
-
- GstFlowReturn ret = GST_FLOW_OK;
-
- GSList *collected;
-
- guint nsamples;
-
- guint ncollected = 0;
-
- gboolean empty = TRUE;
-
- gint width = self->width / 8;
-
- g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED);
- g_return_val_if_fail (self->width > 0, GST_FLOW_NOT_NEGOTIATED);
- g_return_val_if_fail (self->channels > 0, GST_FLOW_NOT_NEGOTIATED);
- g_return_val_if_fail (self->rate > 0, GST_FLOW_NOT_NEGOTIATED);
-
- size = gst_collect_pads_available (pads);
-
- g_return_val_if_fail (size % width == 0, GST_FLOW_ERROR);
-
- GST_DEBUG_OBJECT (self, "Starting to collect %u bytes from %d channels", size,
- self->channels);
-
- nsamples = size / width;
-
- ret =
- gst_pad_alloc_buffer (self->src, GST_BUFFER_OFFSET_NONE,
- size * self->channels, GST_PAD_CAPS (self->src), &outbuf);
-
- if (ret != GST_FLOW_OK) {
- return ret;
- } else if (outbuf == NULL || GST_BUFFER_SIZE (outbuf) < size * self->channels) {
- gst_buffer_unref (outbuf);
- return GST_FLOW_NOT_NEGOTIATED;
- } else if (!gst_caps_is_equal (GST_BUFFER_CAPS (outbuf),
- GST_PAD_CAPS (self->src))) {
- gst_buffer_unref (outbuf);
- return GST_FLOW_NOT_NEGOTIATED;
- }
-
- memset (GST_BUFFER_DATA (outbuf), 0, size * self->channels);
-
- for (collected = pads->data; collected != NULL; collected = collected->next) {
- GstCollectData *cdata;
-
- GstBuffer *inbuf;
-
- guint8 *outdata;
-
- cdata = (GstCollectData *) collected->data;
-
- inbuf = gst_collect_pads_take_buffer (pads, cdata, size);
- if (inbuf == NULL) {
- GST_DEBUG_OBJECT (cdata->pad, "No buffer available");
- goto next;
- }
- ncollected++;
-
- if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP))
- goto next;
-
- empty = FALSE;
- outdata =
- GST_BUFFER_DATA (outbuf) +
- width * GST_INTERLEAVE_PAD_CAST (cdata->pad)->channel;
-
- self->func (outdata, GST_BUFFER_DATA (inbuf), self->channels, nsamples);
-
- next:
- if (inbuf)
- gst_buffer_unref (inbuf);
- }
-
- if (ncollected == 0)
- goto eos;
-
- if (self->segment_pending) {
- GstEvent *event;
-
- event = gst_event_new_new_segment_full (FALSE, self->segment_rate,
- 1.0, GST_FORMAT_TIME, self->timestamp, -1, self->segment_position);
-
- gst_pad_push_event (self->src, event);
- self->segment_pending = FALSE;
- self->segment_position = 0;
- }
-
- GST_BUFFER_TIMESTAMP (outbuf) = self->timestamp;
- GST_BUFFER_OFFSET (outbuf) = self->offset;
-
- self->offset += nsamples;
- self->timestamp = gst_util_uint64_scale_int (self->offset,
- GST_SECOND, self->rate);
-
- GST_BUFFER_DURATION (outbuf) = self->timestamp -
- GST_BUFFER_TIMESTAMP (outbuf);
-
- if (empty)
- GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP);
-
- GST_LOG_OBJECT (self, "pushing outbuf, timestamp %" GST_TIME_FORMAT,
- GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)));
- ret = gst_pad_push (self->src, outbuf);
-
- return ret;
-
-eos:
- {
- GST_DEBUG_OBJECT (self, "no data available, must be EOS");
- gst_buffer_unref (outbuf);
- gst_pad_push_event (self->src, gst_event_new_eos ());
- return GST_FLOW_UNEXPECTED;
- }
-}
diff --git a/gst/interleave/interleave.h b/gst/interleave/interleave.h
deleted file mode 100644
index fb3b2741..00000000
--- a/gst/interleave/interleave.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/* GStreamer
- * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
- * 2000 Wim Taymans <wtay@chello.be>
- * 2005 Wim Taymans <wim@fluendo.com>
- * 2007 Andy Wingo <wingo at pobox.com>
- * 2008 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * interleave.c: interleave samples, mostly based on adder
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 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
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifndef __INTERLEAVE_H__
-#define __INTERLEAVE_H__
-
-#include <gst/gst.h>
-#include <gst/base/gstcollectpads.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_INTERLEAVE (gst_interleave_get_type())
-#define GST_INTERLEAVE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_INTERLEAVE,GstInterleave))
-#define GST_INTERLEAVE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_INTERLEAVE,GstInterleaveClass))
-#define GST_INTERLEAVE_GET_CLASS(obj) \
- (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_INTERLEAVE,GstInterleaveClass))
-#define GST_IS_INTERLEAVE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_INTERLEAVE))
-#define GST_IS_INTERLEAVE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_INTERLEAVE))
-
-typedef struct _GstInterleave GstInterleave;
-typedef struct _GstInterleaveClass GstInterleaveClass;
-
-typedef void (*GstInterleaveFunc) (gpointer out, gpointer in, guint stride, guint nframes);
-
-struct _GstInterleave
-{
- GstElement element;
-
- /*< private >*/
- GstCollectPads *collect;
-
- gint channels;
- gint padcounter;
- gint rate;
- gint width;
-
- GValueArray *channel_positions;
- GValueArray *input_channel_positions;
- gboolean channel_positions_from_input;
-
- GstCaps *sinkcaps;
-
- GstClockTime timestamp;
- guint64 offset;
-
- gboolean segment_pending;
- guint64 segment_position;
- gdouble segment_rate;
- GstSegment segment;
-
- GstPadEventFunction collect_event;
-
- GstInterleaveFunc func;
-
- GstPad *src;
-};
-
-struct _GstInterleaveClass
-{
- GstElementClass parent_class;
-};
-
-GType gst_interleave_get_type (void);
-
-G_END_DECLS
-
-#endif /* __INTERLEAVE_H__ */
diff --git a/gst/interleave/plugin.c b/gst/interleave/plugin.c
deleted file mode 100644
index 7017c45c..00000000
--- a/gst/interleave/plugin.c
+++ /dev/null
@@ -1,44 +0,0 @@
-/* GStreamer interleave plugin
- * Copyright (C) 2004,2007 Andy Wingo <wingo at pobox.com>
- *
- * plugin.c: the stubs for the interleave plugin
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 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
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-#ifdef HAVE_CONFIG_H
-#include "config.h"
-#endif
-
-#include "plugin.h"
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
- if (!gst_element_register (plugin, "interleave",
- GST_RANK_NONE, gst_interleave_get_type ()) ||
- !gst_element_register (plugin, "deinterleave",
- GST_RANK_NONE, gst_deinterleave_get_type ()))
- return FALSE;
-
- return TRUE;
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
- GST_VERSION_MINOR,
- "interleave",
- "Audio interleaver/deinterleaver",
- plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
diff --git a/gst/interleave/plugin.h b/gst/interleave/plugin.h
deleted file mode 100644
index 3e96a7e1..00000000
--- a/gst/interleave/plugin.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/* GStreamer interleave plugin
- * Copyright (C) 2004,2007 Andy Wingo <wingo at pobox.com>
- *
- * plugin.h: the stubs for the interleave plugin
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Library General Public
- * License as published by the Free Software Foundation; either
- * version 2 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
- * Library General Public License for more details.
- *
- * You should have received a copy of the GNU Library General Public
- * License along with this library; if not, write to the
- * Free Software Foundation, Inc., 59 Temple Place - Suite 330,
- * Boston, MA 02111-1307, USA.
- */
-
-
-#ifndef __GST_PLUGIN_INTERLEAVE_H__
-#define __GST_PLUGIN_INTERLEAVE_H__
-
-
-#include <gst/gst.h>
-#include "interleave.h"
-#include "deinterleave.h"
-
-#endif /* __GST_PLUGIN_INTERLEAVE_H__ */
diff --git a/gst/replaygain/Makefile.am b/gst/replaygain/Makefile.am
deleted file mode 100644
index a0a3ca5a..00000000
--- a/gst/replaygain/Makefile.am
+++ /dev/null
@@ -1,21 +0,0 @@
-plugin_LTLIBRARIES = libgstreplaygain.la
-
-libgstreplaygain_la_SOURCES = \
- gstrganalysis.c \
- gstrglimiter.c \
- gstrgvolume.c \
- replaygain.c \
- rganalysis.c
-libgstreplaygain_la_CFLAGS = \
- $(GST_CFLAGS) $(GST_BASE_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
-libgstreplaygain_la_LIBADD = \
- $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstpbutils-0.10 $(LIBM)
-libgstreplaygain_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS)
-
-# headers we need but don't want installed
-noinst_HEADERS = \
- gstrganalysis.h \
- gstrglimiter.h \
- gstrgvolume.h \
- replaygain.h \
- rganalysis.h
diff --git a/gst/replaygain/gstrganalysis.c b/gst/replaygain/gstrganalysis.c
deleted file mode 100644
index 982c8a7f..00000000
--- a/gst/replaygain/gstrganalysis.c
+++ /dev/null
@@ -1,692 +0,0 @@
-/* GStreamer ReplayGain analysis
- *
- * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
- *
- * gstrganalysis.c: Element that performs the ReplayGain analysis
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-/**
- * SECTION:element-rganalysis
- * @see_also: #GstRgVolume
- *
- * This element analyzes raw audio sample data in accordance with the proposed
- * <ulink url="http://replaygain.org">ReplayGain standard</ulink> for
- * calculating the ideal replay gain for music tracks and albums. The element
- * is designed as a pass-through filter that never modifies any data. As it
- * receives an EOS event, it finalizes the ongoing analysis and generates a tag
- * list containing the results. It is sent downstream with a tag event and
- * posted on the message bus with a tag message. The EOS event is forwarded as
- * normal afterwards. Result tag lists at least contain the tags
- * #GST_TAG_TRACK_GAIN, #GST_TAG_TRACK_PEAK and #GST_TAG_REFERENCE_LEVEL.
- *
- * Because the generated metadata tags become available at the end of streams,
- * downstream muxer and encoder elements are normally unable to save them in
- * their output since they generally save metadata in the file header.
- * Therefore, it is often necessary that applications read the results in a bus
- * event handler for the tag message. Obtaining the values this way is always
- * needed for <link linkend="GstRgAnalysis--num-tracks">album processing</link>
- * since the album gain and peak values need to be associated with all tracks of
- * an album, not just the last one.
- *
- * <refsect2>
- * <title>Example launch lines</title>
- * |[
- * gst-launch -t audiotestsrc wave=sine num-buffers=512 ! rganalysis ! fakesink
- * ]| Analyze a simple test waveform
- * |[
- * gst-launch -t filesrc location=filename.ext ! decodebin \
- * ! audioconvert ! audioresample ! rganalysis ! fakesink
- * ]| Analyze a given file
- * |[
- * gst-launch -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav \
- * ! wavparse ! rganalysis ! fakesink
- * ]| Analyze the pink noise reference file
- * <para>
- * The above launch line yields a result gain of +6 dB (instead of the expected
- * +0 dB). This is not in error, refer to the #GstRgAnalysis:reference-level
- * property documentation for more information.
- * </para>
- * </refsect2>
- * <refsect2>
- * <title>Acknowledgements</title>
- * <para>
- * This element is based on code used in the <ulink
- * url="http://sjeng.org/vorbisgain.html">vorbisgain</ulink> program and many
- * others. The relevant parts are copyrighted by David Robinson, Glen Sawyer
- * and Frank Klemm.
- * </para>
- * </refsect2>
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#include <gst/gst.h>
-#include <gst/base/gstbasetransform.h>
-
-#include "gstrganalysis.h"
-#include "replaygain.h"
-
-GST_DEBUG_CATEGORY_STATIC (gst_rg_analysis_debug);
-#define GST_CAT_DEFAULT gst_rg_analysis_debug
-
-static const GstElementDetails rganalysis_details = {
- "ReplayGain analysis",
- "Filter/Analyzer/Audio",
- "Perform the ReplayGain analysis",
- "Ren\xc3\xa9 Stadler <mail@renestadler.de>"
-};
-
-/* Default property value. */
-#define FORCED_DEFAULT TRUE
-
-enum
-{
- PROP_0,
- PROP_NUM_TRACKS,
- PROP_FORCED,
- PROP_REFERENCE_LEVEL
-};
-
-/* The ReplayGain algorithm is intended for use with mono and stereo
- * audio. The used implementation has filter coefficients for the
- * "usual" sample rates in the 8000 to 48000 Hz range. */
-#define REPLAY_GAIN_CAPS \
- "channels = (int) { 1, 2 }, " \
- "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, " \
- "44100, 48000 }"
-
-static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
- "width = (int) 32, " "endianness = (int) BYTE_ORDER, "
- REPLAY_GAIN_CAPS "; "
- "audio/x-raw-int, "
- "width = (int) 16, " "depth = (int) [ 1, 16 ], "
- "signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
- REPLAY_GAIN_CAPS));
-
-static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
- "width = (int) 32, " "endianness = (int) BYTE_ORDER, "
- REPLAY_GAIN_CAPS "; "
- "audio/x-raw-int, "
- "width = (int) 16, " "depth = (int) [ 1, 16 ], "
- "signed = (boolean) true, " "endianness = (int) BYTE_ORDER, "
- REPLAY_GAIN_CAPS));
-
-GST_BOILERPLATE (GstRgAnalysis, gst_rg_analysis, GstBaseTransform,
- GST_TYPE_BASE_TRANSFORM);
-
-static void gst_rg_analysis_class_init (GstRgAnalysisClass * klass);
-static void gst_rg_analysis_init (GstRgAnalysis * filter,
- GstRgAnalysisClass * gclass);
-
-static void gst_rg_analysis_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec);
-static void gst_rg_analysis_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec);
-
-static gboolean gst_rg_analysis_start (GstBaseTransform * base);
-static gboolean gst_rg_analysis_set_caps (GstBaseTransform * base,
- GstCaps * incaps, GstCaps * outcaps);
-static GstFlowReturn gst_rg_analysis_transform_ip (GstBaseTransform * base,
- GstBuffer * buf);
-static gboolean gst_rg_analysis_event (GstBaseTransform * base,
- GstEvent * event);
-static gboolean gst_rg_analysis_stop (GstBaseTransform * base);
-
-static void gst_rg_analysis_handle_tags (GstRgAnalysis * filter,
- const GstTagList * tag_list);
-static void gst_rg_analysis_handle_eos (GstRgAnalysis * filter);
-static gboolean gst_rg_analysis_track_result (GstRgAnalysis * filter,
- GstTagList ** tag_list);
-static gboolean gst_rg_analysis_album_result (GstRgAnalysis * filter,
- GstTagList ** tag_list);
-
-static void
-gst_rg_analysis_base_init (gpointer g_class)
-{
- GstElementClass *element_class = GST_ELEMENT_CLASS (g_class);
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_factory));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&sink_factory));
- gst_element_class_set_details (element_class, &rganalysis_details);
-
- GST_DEBUG_CATEGORY_INIT (gst_rg_analysis_debug, "rganalysis", 0,
- "ReplayGain analysis element");
-}
-
-static void
-gst_rg_analysis_class_init (GstRgAnalysisClass * klass)
-{
- GObjectClass *gobject_class;
- GstBaseTransformClass *trans_class;
-
- gobject_class = (GObjectClass *) klass;
- gobject_class->set_property = gst_rg_analysis_set_property;
- gobject_class->get_property = gst_rg_analysis_get_property;
-
- /**
- * GstRgAnalysis:num-tracks:
- *
- * Number of remaining album tracks.
- *
- * Analyzing several streams sequentially and assigning them a common result
- * gain is known as "album processing". If this gain is used during playback
- * (by switching to "album mode"), all tracks of an album receive the same
- * amplification. This keeps the relative volume levels between the tracks
- * intact. To enable this, set this property to the number of streams that
- * will be processed as album tracks.
- *
- * Every time an EOS event is received, the value of this property is
- * decremented by one. As it reaches zero, it is assumed that the last track
- * of the album finished. The tag list for the final stream will contain the
- * additional tags #GST_TAG_ALBUM_GAIN and #GST_TAG_ALBUM_PEAK. All other
- * streams just get the two track tags posted because the values for the album
- * tags are not known before all tracks are analyzed. Applications need to
- * ensure that the album gain and peak values are also associated with the
- * other tracks when storing the results.
- *
- * If the total number of album tracks is unknown beforehand, just ensure that
- * the value is greater than 1 before each track starts. Then before the end
- * of the last track, set it to the value 1.
- *
- * To perform album processing, the element has to preserve data between
- * streams. This cannot survive a state change to the NULL or READY state.
- * If you change your pipeline's state to NULL or READY between tracks, lock
- * the element's state using gst_element_set_locked_state() when it is in
- * PAUSED or PLAYING.
- */
- g_object_class_install_property (gobject_class, PROP_NUM_TRACKS,
- g_param_spec_int ("num-tracks", "Number of album tracks",
- "Number of remaining album tracks", 0, G_MAXINT, 0,
- G_PARAM_READWRITE));
- /**
- * GstRgAnalysis:forced:
- *
- * Whether to analyze streams even when ReplayGain tags exist.
- *
- * For assisting transcoder/converter applications, the element can silently
- * skip the processing of streams that already contain the necessary tags.
- * Data will flow as usual but the element will not consume CPU time and will
- * not generate result tags. To enable possible skipping, set this property
- * to #FALSE.
- *
- * If used in conjunction with <link linkend="GstRgAnalysis--num-tracks">album
- * processing</link>, the element will skip the number of remaining album
- * tracks if a full set of tags is found for the first track. If a subsequent
- * track of the album is missing tags, processing cannot start again. If this
- * is undesired, the application has to scan all files beforehand and enable
- * forcing of processing if needed.
- */
- g_object_class_install_property (gobject_class, PROP_FORCED,
- g_param_spec_boolean ("forced", "Forced",
- "Analyze even if ReplayGain tags exist",
- FORCED_DEFAULT, G_PARAM_READWRITE));
- /**
- * GstRgAnalysis:reference-level:
- *
- * Reference level [dB].
- *
- * Analyzing the ReplayGain pink noise reference waveform computes a result of
- * +6 dB instead of the expected 0 dB. This is because the default reference
- * level is 89 dB. To obtain values as lined out in the original proposal of
- * ReplayGain, set this property to 83.
- *
- * Almost all software uses 89 dB as a reference however, and this value has
- * become the new official value. That is to say, while the change has been
- * acclaimed by the author of the ReplayGain proposal, the <ulink
- * url="http://replaygain.org">webpage</ulink> is still outdated at the time
- * of this writing.
- *
- * The value was changed because the original proposal recommends a default
- * pre-amp value of +6 dB for playback. This seemed a bit odd, as it means
- * that the algorithm has the general tendency to produce adjustment values
- * that are 6 dB too low. Bumping the reference level by 6 dB compensated for
- * this.
- *
- * The problem of the reference level being ambiguous for lack of concise
- * standardization is to be solved by adopting the #GST_TAG_REFERENCE_LEVEL
- * tag, which allows to store the used value alongside the gain values.
- */
- g_object_class_install_property (gobject_class, PROP_REFERENCE_LEVEL,
- g_param_spec_double ("reference-level", "Reference level",
- "Reference level [dB]", 0.0, 150., RG_REFERENCE_LEVEL,
- G_PARAM_READWRITE));
-
- trans_class = (GstBaseTransformClass *) klass;
- trans_class->start = GST_DEBUG_FUNCPTR (gst_rg_analysis_start);
- trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_rg_analysis_set_caps);
- trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_analysis_transform_ip);
- trans_class->event = GST_DEBUG_FUNCPTR (gst_rg_analysis_event);
- trans_class->stop = GST_DEBUG_FUNCPTR (gst_rg_analysis_stop);
- trans_class->passthrough_on_same_caps = TRUE;
-}
-
-static void
-gst_rg_analysis_init (GstRgAnalysis * filter, GstRgAnalysisClass * gclass)
-{
- GstBaseTransform *base = GST_BASE_TRANSFORM (filter);
-
- gst_base_transform_set_gap_aware (base, TRUE);
-
- filter->num_tracks = 0;
- filter->forced = FORCED_DEFAULT;
- filter->reference_level = RG_REFERENCE_LEVEL;
-
- filter->ctx = NULL;
- filter->analyze = NULL;
-}
-
-static void
-gst_rg_analysis_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
- GstRgAnalysis *filter = GST_RG_ANALYSIS (object);
-
- switch (prop_id) {
- case PROP_NUM_TRACKS:
- filter->num_tracks = g_value_get_int (value);
- break;
- case PROP_FORCED:
- filter->forced = g_value_get_boolean (value);
- break;
- case PROP_REFERENCE_LEVEL:
- filter->reference_level = g_value_get_double (value);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static void
-gst_rg_analysis_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec)
-{
- GstRgAnalysis *filter = GST_RG_ANALYSIS (object);
-
- switch (prop_id) {
- case PROP_NUM_TRACKS:
- g_value_set_int (value, filter->num_tracks);
- break;
- case PROP_FORCED:
- g_value_set_boolean (value, filter->forced);
- break;
- case PROP_REFERENCE_LEVEL:
- g_value_set_double (value, filter->reference_level);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static gboolean
-gst_rg_analysis_start (GstBaseTransform * base)
-{
- GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
-
- filter->ignore_tags = FALSE;
- filter->skip = FALSE;
- filter->has_track_gain = FALSE;
- filter->has_track_peak = FALSE;
- filter->has_album_gain = FALSE;
- filter->has_album_peak = FALSE;
-
- filter->ctx = rg_analysis_new ();
- filter->analyze = NULL;
-
- GST_LOG_OBJECT (filter, "started");
-
- return TRUE;
-}
-
-static gboolean
-gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps,
- GstCaps * out_caps)
-{
- GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
- GstStructure *structure;
- const gchar *name;
- gint n_channels, sample_rate, sample_bit_size, sample_size;
-
- g_return_val_if_fail (filter->ctx != NULL, FALSE);
-
- GST_DEBUG_OBJECT (filter,
- "set_caps in %" GST_PTR_FORMAT " out %" GST_PTR_FORMAT,
- in_caps, out_caps);
-
- structure = gst_caps_get_structure (in_caps, 0);
- name = gst_structure_get_name (structure);
-
- if (!gst_structure_get_int (structure, "width", &sample_bit_size)
- || !gst_structure_get_int (structure, "channels", &n_channels)
- || !gst_structure_get_int (structure, "rate", &sample_rate))
- goto invalid_format;
-
- if (!rg_analysis_set_sample_rate (filter->ctx, sample_rate))
- goto invalid_format;
-
- if (sample_bit_size % 8 != 0)
- goto invalid_format;
- sample_size = sample_bit_size / 8;
-
- if (g_str_equal (name, "audio/x-raw-float")) {
-
- if (sample_size != sizeof (gfloat))
- goto invalid_format;
-
- /* The depth is not variable for float formats of course. It just
- * makes the transform function nice and simple if the
- * rg_analysis_analyze_* functions have a common signature. */
- filter->depth = sizeof (gfloat) * 8;
-
- if (n_channels == 1)
- filter->analyze = rg_analysis_analyze_mono_float;
- else if (n_channels == 2)
- filter->analyze = rg_analysis_analyze_stereo_float;
- else
- goto invalid_format;
-
- } else if (g_str_equal (name, "audio/x-raw-int")) {
-
- if (sample_size != sizeof (gint16))
- goto invalid_format;
-
- if (!gst_structure_get_int (structure, "depth", &filter->depth))
- goto invalid_format;
- if (filter->depth < 1 || filter->depth > 16)
- goto invalid_format;
-
- if (n_channels == 1)
- filter->analyze = rg_analysis_analyze_mono_int16;
- else if (n_channels == 2)
- filter->analyze = rg_analysis_analyze_stereo_int16;
- else
- goto invalid_format;
-
- } else {
-
- goto invalid_format;
- }
-
- return TRUE;
-
- /* Errors. */
-invalid_format:
- {
- filter->analyze = NULL;
- GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION,
- ("Invalid incoming caps: %" GST_PTR_FORMAT, in_caps), (NULL));
- return FALSE;
- }
-}
-
-static GstFlowReturn
-gst_rg_analysis_transform_ip (GstBaseTransform * base, GstBuffer * buf)
-{
- GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
-
- g_return_val_if_fail (filter->ctx != NULL, GST_FLOW_WRONG_STATE);
- g_return_val_if_fail (filter->analyze != NULL, GST_FLOW_NOT_NEGOTIATED);
-
- if (filter->skip)
- return GST_FLOW_OK;
-
- /* Buffers made up of silence have no influence on the analysis: */
- if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP))
- return GST_FLOW_OK;
-
- GST_LOG_OBJECT (filter, "processing buffer of size %u",
- GST_BUFFER_SIZE (buf));
-
- filter->analyze (filter->ctx, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf),
- filter->depth);
-
- return GST_FLOW_OK;
-}
-
-static gboolean
-gst_rg_analysis_event (GstBaseTransform * base, GstEvent * event)
-{
- GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
-
- g_return_val_if_fail (filter->ctx != NULL, TRUE);
-
- switch (GST_EVENT_TYPE (event)) {
-
- case GST_EVENT_EOS:
- {
- GST_LOG_OBJECT (filter, "received EOS event");
-
- gst_rg_analysis_handle_eos (filter);
-
- GST_LOG_OBJECT (filter, "passing on EOS event");
-
- break;
- }
- case GST_EVENT_TAG:
- {
- GstTagList *tag_list;
-
- /* The reference to the tag list is borrowed. */
- gst_event_parse_tag (event, &tag_list);
- gst_rg_analysis_handle_tags (filter, tag_list);
-
- break;
- }
- default:
- break;
- }
-
- return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event);
-}
-
-static gboolean
-gst_rg_analysis_stop (GstBaseTransform * base)
-{
- GstRgAnalysis *filter = GST_RG_ANALYSIS (base);
-
- g_return_val_if_fail (filter->ctx != NULL, FALSE);
-
- rg_analysis_destroy (filter->ctx);
- filter->ctx = NULL;
-
- GST_LOG_OBJECT (filter, "stopped");
-
- return TRUE;
-}
-
-static void
-gst_rg_analysis_handle_tags (GstRgAnalysis * filter,
- const GstTagList * tag_list)
-{
- gboolean album_processing = (filter->num_tracks > 0);
- gdouble dummy;
-
- if (!album_processing)
- filter->ignore_tags = FALSE;
-
- if (filter->skip && album_processing) {
- GST_DEBUG_OBJECT (filter, "ignoring tag event: skipping album");
- return;
- } else if (filter->skip) {
- GST_DEBUG_OBJECT (filter, "ignoring tag event: skipping track");
- return;
- } else if (filter->ignore_tags) {
- GST_DEBUG_OBJECT (filter, "ignoring tag event: cannot skip anyways");
- return;
- }
-
- filter->has_track_gain |= gst_tag_list_get_double (tag_list,
- GST_TAG_TRACK_GAIN, &dummy);
- filter->has_track_peak |= gst_tag_list_get_double (tag_list,
- GST_TAG_TRACK_PEAK, &dummy);
- filter->has_album_gain |= gst_tag_list_get_double (tag_list,
- GST_TAG_ALBUM_GAIN, &dummy);
- filter->has_album_peak |= gst_tag_list_get_double (tag_list,
- GST_TAG_ALBUM_PEAK, &dummy);
-
- if (!(filter->has_track_gain && filter->has_track_peak)) {
- GST_DEBUG_OBJECT (filter, "track tags not complete yet");
- return;
- }
-
- if (album_processing && !(filter->has_album_gain && filter->has_album_peak)) {
- GST_DEBUG_OBJECT (filter, "album tags not complete yet");
- return;
- }
-
- if (filter->forced) {
- GST_DEBUG_OBJECT (filter,
- "existing tags are sufficient, but processing anyway (forced)");
- return;
- }
-
- filter->skip = TRUE;
- rg_analysis_reset (filter->ctx);
-
- if (!album_processing) {
- GST_DEBUG_OBJECT (filter,
- "existing tags are sufficient, will not process this track");
- } else {
- GST_DEBUG_OBJECT (filter,
- "existing tags are sufficient, will not process this album");
- }
-}
-
-static void
-gst_rg_analysis_handle_eos (GstRgAnalysis * filter)
-{
- gboolean album_processing = (filter->num_tracks > 0);
- gboolean album_finished = (filter->num_tracks == 1);
- gboolean album_skipping = album_processing && filter->skip;
-
- filter->has_track_gain = FALSE;
- filter->has_track_peak = FALSE;
-
- if (album_finished) {
- filter->ignore_tags = FALSE;
- filter->skip = FALSE;
- filter->has_album_gain = FALSE;
- filter->has_album_peak = FALSE;
- } else if (!album_skipping) {
- filter->skip = FALSE;
- }
-
- /* We might have just fully processed a track because it has
- * incomplete tags. If we do album processing and allow skipping
- * (not forced), prevent switching to skipping if a later track with
- * full tags comes along: */
- if (!filter->forced && album_processing && !album_finished)
- filter->ignore_tags = TRUE;
-
- if (!filter->skip) {
- GstTagList *tag_list = NULL;
- gboolean track_success;
- gboolean album_success = FALSE;
-
- track_success = gst_rg_analysis_track_result (filter, &tag_list);
-
- if (album_finished)
- album_success = gst_rg_analysis_album_result (filter, &tag_list);
- else if (!album_processing)
- rg_analysis_reset_album (filter->ctx);
-
- if (track_success || album_success) {
- GST_LOG_OBJECT (filter, "posting tag list with results");
- gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
- GST_TAG_REFERENCE_LEVEL, filter->reference_level, NULL);
- /* This steals our reference to the list: */
- gst_element_found_tags_for_pad (GST_ELEMENT (filter),
- GST_BASE_TRANSFORM_SRC_PAD (GST_BASE_TRANSFORM (filter)), tag_list);
- }
- }
-
- if (album_processing) {
- filter->num_tracks--;
-
- if (!album_finished) {
- GST_DEBUG_OBJECT (filter, "album not finished yet (num-tracks is now %u)",
- filter->num_tracks);
- } else {
- GST_DEBUG_OBJECT (filter, "album finished (num-tracks is now 0)");
- }
- }
-
- if (album_processing)
- g_object_notify (G_OBJECT (filter), "num-tracks");
-}
-
-static gboolean
-gst_rg_analysis_track_result (GstRgAnalysis * filter, GstTagList ** tag_list)
-{
- gboolean track_success;
- gdouble track_gain, track_peak;
-
- track_success = rg_analysis_track_result (filter->ctx, &track_gain,
- &track_peak);
-
- if (track_success) {
- track_gain += filter->reference_level - RG_REFERENCE_LEVEL;
- GST_INFO_OBJECT (filter, "track gain is %+.2f dB, peak %.6f", track_gain,
- track_peak);
- } else {
- GST_INFO_OBJECT (filter, "track was too short to analyze");
- }
-
- if (track_success) {
- if (*tag_list == NULL)
- *tag_list = gst_tag_list_new ();
- gst_tag_list_add (*tag_list, GST_TAG_MERGE_APPEND,
- GST_TAG_TRACK_PEAK, track_peak, GST_TAG_TRACK_GAIN, track_gain, NULL);
- }
-
- return track_success;
-}
-
-static gboolean
-gst_rg_analysis_album_result (GstRgAnalysis * filter, GstTagList ** tag_list)
-{
- gboolean album_success;
- gdouble album_gain, album_peak;
-
- album_success = rg_analysis_album_result (filter->ctx, &album_gain,
- &album_peak);
-
- if (album_success) {
- album_gain += filter->reference_level - RG_REFERENCE_LEVEL;
- GST_INFO_OBJECT (filter, "album gain is %+.2f dB, peak %.6f", album_gain,
- album_peak);
- } else {
- GST_INFO_OBJECT (filter, "album was too short to analyze");
- }
-
- if (album_success) {
- if (*tag_list == NULL)
- *tag_list = gst_tag_list_new ();
- gst_tag_list_add (*tag_list, GST_TAG_MERGE_APPEND,
- GST_TAG_ALBUM_PEAK, album_peak, GST_TAG_ALBUM_GAIN, album_gain, NULL);
- }
-
- return album_success;
-}
diff --git a/gst/replaygain/gstrganalysis.h b/gst/replaygain/gstrganalysis.h
deleted file mode 100644
index fbf46830..00000000
--- a/gst/replaygain/gstrganalysis.h
+++ /dev/null
@@ -1,85 +0,0 @@
-/* GStreamer ReplayGain analysis
- *
- * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
- *
- * gstrganalysis.h: Element that performs the ReplayGain analysis
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-#ifndef __GST_RG_ANALYSIS_H__
-#define __GST_RG_ANALYSIS_H__
-
-#include <gst/gst.h>
-#include <gst/base/gstbasetransform.h>
-
-#include "rganalysis.h"
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_RG_ANALYSIS \
- (gst_rg_analysis_get_type())
-#define GST_RG_ANALYSIS(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_ANALYSIS,GstRgAnalysis))
-#define GST_RG_ANALYSIS_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_ANALYSIS,GstRgAnalysisClass))
-#define GST_IS_RG_ANALYSIS(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_ANALYSIS))
-#define GST_IS_RG_ANALYSIS_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_ANALYSIS))
-typedef struct _GstRgAnalysis GstRgAnalysis;
-typedef struct _GstRgAnalysisClass GstRgAnalysisClass;
-
-/**
- * GstRgAnalysis:
- *
- * Opaque data structure.
- */
-struct _GstRgAnalysis
-{
- GstBaseTransform element;
-
- /*< private >*/
-
- RgAnalysisCtx *ctx;
- void (*analyze) (RgAnalysisCtx * ctx, gconstpointer data, gsize size,
- guint depth);
- gint depth;
-
- /* Property values. */
- guint num_tracks;
- gdouble reference_level;
- gboolean forced;
-
- /* State machinery for skipping. */
- gboolean ignore_tags;
- gboolean skip;
- gboolean has_track_gain;
- gboolean has_track_peak;
- gboolean has_album_gain;
- gboolean has_album_peak;
-};
-
-struct _GstRgAnalysisClass
-{
- GstBaseTransformClass parent_class;
-};
-
-GType gst_rg_analysis_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_RG_ANALYSIS_H__ */
diff --git a/gst/replaygain/gstrglimiter.c b/gst/replaygain/gstrglimiter.c
deleted file mode 100644
index 43c7b01a..00000000
--- a/gst/replaygain/gstrglimiter.c
+++ /dev/null
@@ -1,202 +0,0 @@
-/* GStreamer ReplayGain limiter
- *
- * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
- *
- * gstrglimiter.c: Element to apply signal compression to raw audio data
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-/**
- * SECTION:element-rglimiter
- * @see_also: #GstRgVolume
- *
- * This element applies signal compression/limiting to raw audio data. It
- * performs strict hard limiting with soft-knee characteristics, using a
- * threshold of -6 dB. This type of filter is mentioned in the proposed <ulink
- * url="http://replaygain.org">ReplayGain standard</ulink>.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch filesrc location=filename.ext ! decodebin ! audioconvert \
- * ! rgvolume pre-amp=6.0 headroom=10.0 ! rglimiter \
- * ! audioconvert ! audioresample ! alsasink
- * ]|Playback of a file
- * </refsect2>
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#include <gst/gst.h>
-#include <math.h>
-
-#include "gstrglimiter.h"
-
-GST_DEBUG_CATEGORY_STATIC (gst_rg_limiter_debug);
-#define GST_CAT_DEFAULT gst_rg_limiter_debug
-
-enum
-{
- PROP_0,
- PROP_ENABLED,
-};
-
-static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
- "width = (int) 32, channels = (int) [1, MAX], "
- "rate = (int) [1, MAX], endianness = (int) BYTE_ORDER"));
-
-static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
- "width = (int) 32, channels = (int) [1, MAX], "
- "rate = (int) [1, MAX], endianness = (int) BYTE_ORDER"));
-
-GST_BOILERPLATE (GstRgLimiter, gst_rg_limiter, GstBaseTransform,
- GST_TYPE_BASE_TRANSFORM);
-
-static void gst_rg_limiter_class_init (GstRgLimiterClass * klass);
-static void gst_rg_limiter_init (GstRgLimiter * filter,
- GstRgLimiterClass * gclass);
-
-static void gst_rg_limiter_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec);
-static void gst_rg_limiter_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec);
-
-static GstFlowReturn gst_rg_limiter_transform_ip (GstBaseTransform * base,
- GstBuffer * buf);
-
-static const GstElementDetails element_details = {
- "ReplayGain limiter",
- "Filter/Effect/Audio",
- "Apply signal compression to raw audio data",
- "Ren\xc3\xa9 Stadler <mail@renestadler.de>"
-};
-
-static void
-gst_rg_limiter_base_init (gpointer g_class)
-{
- GstElementClass *element_class = g_class;
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_factory));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&sink_factory));
- gst_element_class_set_details (element_class, &element_details);
-
- GST_DEBUG_CATEGORY_INIT (gst_rg_limiter_debug, "rglimiter", 0,
- "ReplayGain limiter element");
-}
-
-static void
-gst_rg_limiter_class_init (GstRgLimiterClass * klass)
-{
- GObjectClass *gobject_class;
- GstBaseTransformClass *trans_class;
-
- gobject_class = (GObjectClass *) klass;
-
- gobject_class->set_property = gst_rg_limiter_set_property;
- gobject_class->get_property = gst_rg_limiter_get_property;
-
- g_object_class_install_property (gobject_class, PROP_ENABLED,
- g_param_spec_boolean ("enabled", "Enabled", "Enable processing", TRUE,
- G_PARAM_READWRITE));
-
- trans_class = GST_BASE_TRANSFORM_CLASS (klass);
- trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_limiter_transform_ip);
- trans_class->passthrough_on_same_caps = FALSE;
-}
-
-static void
-gst_rg_limiter_init (GstRgLimiter * filter, GstRgLimiterClass * gclass)
-{
- GstBaseTransform *base = GST_BASE_TRANSFORM (filter);
-
- gst_base_transform_set_passthrough (base, FALSE);
- gst_base_transform_set_gap_aware (base, TRUE);
-
- filter->enabled = TRUE;
-}
-
-static void
-gst_rg_limiter_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
- GstRgLimiter *filter = GST_RG_LIMITER (object);
-
- switch (prop_id) {
- case PROP_ENABLED:
- filter->enabled = g_value_get_boolean (value);
- gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter),
- !filter->enabled);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static void
-gst_rg_limiter_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec)
-{
- GstRgLimiter *filter = GST_RG_LIMITER (object);
-
- switch (prop_id) {
- case PROP_ENABLED:
- g_value_set_boolean (value, filter->enabled);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-#define LIMIT 1.0
-#define THRES 0.5 /* ca. -6 dB */
-#define COMPL 0.5 /* LIMIT - THRESH */
-
-static GstFlowReturn
-gst_rg_limiter_transform_ip (GstBaseTransform * base, GstBuffer * buf)
-{
- GstRgLimiter *filter = GST_RG_LIMITER (base);
- gfloat *input;
- guint count;
- guint i;
-
- if (!filter->enabled)
- return GST_FLOW_OK;
-
- if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP))
- return GST_FLOW_OK;
-
- input = (gfloat *) GST_BUFFER_DATA (buf);
- count = GST_BUFFER_SIZE (buf) / sizeof (gfloat);
-
- for (i = count; i--;) {
- if (*input > THRES)
- *input = tanhf ((*input - THRES) / COMPL) * COMPL + THRES;
- else if (*input < -THRES)
- *input = tanhf ((*input + THRES) / COMPL) * COMPL - THRES;
- input++;
- }
-
- return GST_FLOW_OK;
-}
diff --git a/gst/replaygain/gstrglimiter.h b/gst/replaygain/gstrglimiter.h
deleted file mode 100644
index 63bd8049..00000000
--- a/gst/replaygain/gstrglimiter.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/* GStreamer ReplayGain limiter
- *
- * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
- *
- * gstrglimiter.h: Element to apply signal compression to raw audio data
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-#ifndef __GST_RG_LIMITER_H__
-#define __GST_RG_LIMITER_H__
-
-#include <gst/gst.h>
-#include <gst/base/gstbasetransform.h>
-
-#define GST_TYPE_RG_LIMITER \
- (gst_rg_limiter_get_type())
-#define GST_RG_LIMITER(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_LIMITER,GstRgLimiter))
-#define GST_RG_LIMITER_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_LIMITER,GstRgLimiterClass))
-#define GST_IS_RG_LIMITER(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_LIMITER))
-#define GST_IS_RG_LIMITER_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_LIMITER))
-
-typedef struct _GstRgLimiter GstRgLimiter;
-typedef struct _GstRgLimiterClass GstRgLimiterClass;
-
-/**
- * GstRgLimiter:
- *
- * Opaque data structure.
- */
-struct _GstRgLimiter
-{
- GstBaseTransform element;
-
- /*< private >*/
-
- gboolean enabled;
-};
-
-struct _GstRgLimiterClass
-{
- GstBaseTransformClass parent_class;
-};
-
-GType gst_rg_limiter_get_type (void);
-
-#endif /* __GST_RG_LIMITER_H__ */
diff --git a/gst/replaygain/gstrgvolume.c b/gst/replaygain/gstrgvolume.c
deleted file mode 100644
index 41fe441d..00000000
--- a/gst/replaygain/gstrgvolume.c
+++ /dev/null
@@ -1,698 +0,0 @@
-/* GStreamer ReplayGain volume adjustment
- *
- * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
- *
- * gstrgvolume.c: Element to apply ReplayGain volume adjustment
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-/**
- * SECTION:element-rgvolume
- * @see_also: #GstRgLimiter, #GstRgAnalysis
- *
- * This element applies volume changes to streams as lined out in the proposed
- * <ulink url="http://replaygain.org">ReplayGain standard</ulink>. It
- * interprets the ReplayGain meta data tags and carries out the adjustment (by
- * using a volume element internally). The relevant tags are:
- * <itemizedlist>
- * <listitem>#GST_TAG_TRACK_GAIN</listitem>
- * <listitem>#GST_TAG_TRACK_PEAK</listitem>
- * <listitem>#GST_TAG_ALBUM_GAIN</listitem>
- * <listitem>#GST_TAG_ALBUM_PEAK</listitem>
- * <listitem>#GST_TAG_REFERENCE_LEVEL</listitem>
- * </itemizedlist>
- * The information carried by these tags must have been calculated beforehand by
- * performing the ReplayGain analysis. This is implemented by the <link
- * linkend="GstRgAnalysis">rganalysis</link> element.
- *
- * The signal compression/limiting recommendations outlined in the proposed
- * standard are not implemented by this element. This has to be handled by
- * separate elements because applications might want to have additional filters
- * between the volume adjustment and the limiting stage. A basic limiter is
- * included with this plugin: The <link linkend="GstRgLimiter">rglimiter</link>
- * element applies -6 dB hard limiting as mentioned in the ReplayGain standard.
- *
- * <refsect2>
- * <title>Example launch line</title>
- * |[
- * gst-launch filesrc location=filename.ext ! decodebin ! audioconvert \
- * ! rgvolume ! audioconvert ! audioresample ! alsasink
- * ]| Playback of a file
- * </refsect2>
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#include <gst/gst.h>
-#include <gst/pbutils/pbutils.h>
-#include <math.h>
-
-#include "gstrgvolume.h"
-#include "replaygain.h"
-
-GST_DEBUG_CATEGORY_STATIC (gst_rg_volume_debug);
-#define GST_CAT_DEFAULT gst_rg_volume_debug
-
-enum
-{
- PROP_0,
- PROP_ALBUM_MODE,
- PROP_HEADROOM,
- PROP_PRE_AMP,
- PROP_FALLBACK_GAIN,
- PROP_TARGET_GAIN,
- PROP_RESULT_GAIN
-};
-
-#define DEFAULT_ALBUM_MODE TRUE
-#define DEFAULT_HEADROOM 0.0
-#define DEFAULT_PRE_AMP 0.0
-#define DEFAULT_FALLBACK_GAIN 0.0
-
-#define DB_TO_LINEAR(x) pow (10., (x) / 20.)
-#define LINEAR_TO_DB(x) (20. * log10 (x))
-
-#define GAIN_FORMAT "+.02f dB"
-#define PEAK_FORMAT ".06f"
-
-#define VALID_GAIN(x) ((x) > -60.00 && (x) < 60.00)
-#define VALID_PEAK(x) ((x) > 0.)
-
-/* Same template caps as GstVolume, for I don't like having just ANY caps. */
-
-static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, MAX ], "
- "endianness = (int) BYTE_ORDER, "
- "width = (int) 32; "
- "audio/x-raw-int, "
- "channels = (int) [ 1, MAX ], "
- "rate = (int) [ 1, MAX ], "
- "endianness = (int) BYTE_ORDER, "
- "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE"));
-
-static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, "
- "rate = (int) [ 1, MAX ], "
- "channels = (int) [ 1, MAX ], "
- "endianness = (int) BYTE_ORDER, "
- "width = (int) 32; "
- "audio/x-raw-int, "
- "channels = (int) [ 1, MAX ], "
- "rate = (int) [ 1, MAX ], "
- "endianness = (int) BYTE_ORDER, "
- "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE"));
-
-GST_BOILERPLATE (GstRgVolume, gst_rg_volume, GstBin, GST_TYPE_BIN);
-
-static void gst_rg_volume_class_init (GstRgVolumeClass * klass);
-static void gst_rg_volume_init (GstRgVolume * self, GstRgVolumeClass * gclass);
-
-static void gst_rg_volume_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec);
-static void gst_rg_volume_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec);
-static void gst_rg_volume_dispose (GObject * object);
-
-static GstStateChangeReturn gst_rg_volume_change_state (GstElement * element,
- GstStateChange transition);
-static gboolean gst_rg_volume_sink_event (GstPad * pad, GstEvent * event);
-
-static GstEvent *gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event);
-static void gst_rg_volume_reset (GstRgVolume * self);
-static void gst_rg_volume_update_gain (GstRgVolume * self);
-static inline void gst_rg_volume_determine_gain (GstRgVolume * self,
- gdouble * target_gain, gdouble * result_gain);
-
-static void
-gst_rg_volume_base_init (gpointer g_class)
-{
- GstElementClass *element_class = g_class;
-
- static const GstElementDetails element_details = {
- "ReplayGain volume",
- "Filter/Effect/Audio",
- "Apply ReplayGain volume adjustment",
- "Ren\xc3\xa9 Stadler <mail@renestadler.de>"
- };
-
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&src_template));
- gst_element_class_add_pad_template (element_class,
- gst_static_pad_template_get (&sink_template));
- gst_element_class_set_details (element_class, &element_details);
-
- GST_DEBUG_CATEGORY_INIT (gst_rg_volume_debug, "rgvolume", 0,
- "ReplayGain volume element");
-}
-
-static void
-gst_rg_volume_class_init (GstRgVolumeClass * klass)
-{
- GObjectClass *gobject_class;
- GstElementClass *element_class;
- GstBinClass *bin_class;
-
- gobject_class = (GObjectClass *) klass;
-
- gobject_class->set_property = gst_rg_volume_set_property;
- gobject_class->get_property = gst_rg_volume_get_property;
- gobject_class->dispose = gst_rg_volume_dispose;
-
- /**
- * GstRgVolume:album-mode:
- *
- * Whether to prefer album gain over track gain.
- *
- * If set to %TRUE, use album gain instead of track gain if both are
- * available. This keeps the relative loudness levels of tracks from the same
- * album intact.
- *
- * If set to %FALSE, track mode is used instead. This effectively leads to
- * more extensive normalization.
- *
- * If album mode is enabled but the album gain tag is absent in the stream,
- * the track gain is used instead. If both gain tags are missing, the value
- * of the <link linkend="GstRgVolume--fallback-gain">fallback-gain</link>
- * property is used instead.
- */
- g_object_class_install_property (gobject_class, PROP_ALBUM_MODE,
- g_param_spec_boolean ("album-mode", "Album mode",
- "Prefer album over track gain", DEFAULT_ALBUM_MODE,
- G_PARAM_READWRITE));
- /**
- * GstRgVolume:headroom:
- *
- * Extra headroom [dB]. This controls the amount by which the output can
- * exceed digital full scale.
- *
- * Only set this to a value greater than 0.0 if signal compression/limiting of
- * a suitable form is applied to the output (or output is brought into the
- * correct range by some other transformation).
- *
- * This element internally uses a volume element, which also supports
- * operating on integer audio formats. These formats do not allow exceeding
- * digital full scale. If extra headroom is used, make sure that the raw
- * audio data format is floating point (audio/x-raw-float). Otherwise,
- * clipping distortion might be introduced as part of the volume adjustment
- * itself.
- */
- g_object_class_install_property (gobject_class, PROP_HEADROOM,
- g_param_spec_double ("headroom", "Headroom", "Extra headroom [dB]",
- 0., 60., DEFAULT_HEADROOM, G_PARAM_READWRITE));
- /**
- * GstRgVolume:pre-amp:
- *
- * Additional gain to apply globally [dB]. This controls the trade-off
- * between uniformity of normalization and utilization of available dynamic
- * range.
- *
- * Note that the default value is 0 dB because the ReplayGain reference value
- * was adjusted by +6 dB (from 83 to 89 dB). At the time of this writing, the
- * <ulink url="http://replaygain.org">webpage</ulink> is still outdated and
- * does not reflect this change however. Where the original proposal states
- * that a proper default pre-amp value is +6 dB, this translates to the used 0
- * dB.
- */
- g_object_class_install_property (gobject_class, PROP_PRE_AMP,
- g_param_spec_double ("pre-amp", "Pre-amp", "Extra gain [dB]",
- -60., 60., DEFAULT_PRE_AMP, G_PARAM_READWRITE));
- /**
- * GstRgVolume:fallback-gain:
- *
- * Fallback gain [dB] for streams missing ReplayGain tags.
- */
- g_object_class_install_property (gobject_class, PROP_FALLBACK_GAIN,
- g_param_spec_double ("fallback-gain", "Fallback gain",
- "Gain for streams missing tags [dB]",
- -60., 60., DEFAULT_FALLBACK_GAIN, G_PARAM_READWRITE));
- /**
- * GstRgVolume:result-gain:
- *
- * Applied gain [dB]. This gain is applied to processed buffer data.
- *
- * This is set to the <link linkend="GstRgVolume--target-gain">target
- * gain</link> if amplification by that amount can be applied safely.
- * "Safely" means that the volume adjustment does not inflict clipping
- * distortion. Should this not be the case, the result gain is set to an
- * appropriately reduced value (by applying peak normalization). The proposed
- * standard calls this "clipping prevention".
- *
- * The difference between target and result gain reflects the necessary amount
- * of reduction. Applications can make use of this information to temporarily
- * reduce the <link linkend="GstRgVolume--pre-amp">pre-amp</link> for
- * subsequent streams, as recommended by the ReplayGain standard.
- *
- * Note that target and result gain differing for a great majority of streams
- * indicates a problem: What happens in this case is that most streams receive
- * peak normalization instead of amplification by the ideal replay gain. To
- * prevent this, the <link linkend="GstRgVolume--pre-amp">pre-amp</link> has
- * to be lowered and/or a limiter has to be used which facilitates the use of
- * <link linkend="GstRgVolume--headroom">headroom</link>.
- */
- g_object_class_install_property (gobject_class, PROP_RESULT_GAIN,
- g_param_spec_double ("result-gain", "Result-gain", "Applied gain [dB]",
- -120., 120., 0., G_PARAM_READABLE));
- /**
- * GstRgVolume:target-gain:
- *
- * Applicable gain [dB]. This gain is supposed to be applied.
- *
- * Depending on the value of the <link
- * linkend="GstRgVolume--album-mode">album-mode</link> property and the
- * presence of ReplayGain tags in the stream, this is set according to one of
- * these simple formulas:
- *
- * <itemizedlist>
- * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + album gain
- * of the stream</listitem>
- * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + track gain
- * of the stream</listitem>
- * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + <link
- * linkend="GstRgVolume--fallback-gain">fallback gain</link></listitem>
- * </itemizedlist>
- */
- g_object_class_install_property (gobject_class, PROP_TARGET_GAIN,
- g_param_spec_double ("target-gain", "Target-gain",
- "Applicable gain [dB]", -120., 120., 0., G_PARAM_READABLE));
-
- element_class = (GstElementClass *) klass;
- element_class->change_state = GST_DEBUG_FUNCPTR (gst_rg_volume_change_state);
-
- bin_class = (GstBinClass *) klass;
- /* Setting these to NULL makes gst_bin_add and _remove refuse to let anyone
- * mess with our internals. */
- bin_class->add_element = NULL;
- bin_class->remove_element = NULL;
-}
-
-static void
-gst_rg_volume_init (GstRgVolume * self, GstRgVolumeClass * gclass)
-{
- GObjectClass *volume_class;
- GstPad *volume_pad, *ghost_pad;
-
- self->album_mode = DEFAULT_ALBUM_MODE;
- self->headroom = DEFAULT_HEADROOM;
- self->pre_amp = DEFAULT_PRE_AMP;
- self->fallback_gain = DEFAULT_FALLBACK_GAIN;
- self->target_gain = 0.0;
- self->result_gain = 0.0;
-
- self->volume_element = gst_element_factory_make ("volume", "rgvolume-volume");
- if (G_UNLIKELY (self->volume_element == NULL)) {
- GstMessage *msg;
-
- GST_WARNING_OBJECT (self, "could not create volume element");
- msg = gst_missing_element_message_new (GST_ELEMENT_CAST (self), "volume");
- gst_element_post_message (GST_ELEMENT_CAST (self), msg);
-
- /* Nothing else to do, we will refuse the state change from NULL to READY to
- * indicate that something went very wrong. It is doubtful that someone
- * attempts changing our state though, since we end up having no pads! */
- return;
- }
-
- volume_class = G_OBJECT_GET_CLASS (G_OBJECT (self->volume_element));
- self->max_volume = G_PARAM_SPEC_DOUBLE
- (g_object_class_find_property (volume_class, "volume"))->maximum;
-
- GST_BIN_CLASS (parent_class)->add_element (GST_BIN_CAST (self),
- self->volume_element);
-
- volume_pad = gst_element_get_static_pad (self->volume_element, "sink");
- ghost_pad = gst_ghost_pad_new_from_template ("sink", volume_pad,
- gst_pad_get_pad_template (volume_pad));
- gst_object_unref (volume_pad);
- gst_pad_set_event_function (ghost_pad, gst_rg_volume_sink_event);
- gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad);
-
- volume_pad = gst_element_get_static_pad (self->volume_element, "src");
- ghost_pad = gst_ghost_pad_new_from_template ("src", volume_pad,
- gst_pad_get_pad_template (volume_pad));
- gst_object_unref (volume_pad);
- gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad);
-}
-
-static void
-gst_rg_volume_set_property (GObject * object, guint prop_id,
- const GValue * value, GParamSpec * pspec)
-{
- GstRgVolume *self = GST_RG_VOLUME (object);
-
- switch (prop_id) {
- case PROP_ALBUM_MODE:
- self->album_mode = g_value_get_boolean (value);
- break;
- case PROP_HEADROOM:
- self->headroom = g_value_get_double (value);
- break;
- case PROP_PRE_AMP:
- self->pre_amp = g_value_get_double (value);
- break;
- case PROP_FALLBACK_GAIN:
- self->fallback_gain = g_value_get_double (value);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-
- gst_rg_volume_update_gain (self);
-}
-
-static void
-gst_rg_volume_get_property (GObject * object, guint prop_id,
- GValue * value, GParamSpec * pspec)
-{
- GstRgVolume *self = GST_RG_VOLUME (object);
-
- switch (prop_id) {
- case PROP_ALBUM_MODE:
- g_value_set_boolean (value, self->album_mode);
- break;
- case PROP_HEADROOM:
- g_value_set_double (value, self->headroom);
- break;
- case PROP_PRE_AMP:
- g_value_set_double (value, self->pre_amp);
- break;
- case PROP_FALLBACK_GAIN:
- g_value_set_double (value, self->fallback_gain);
- break;
- case PROP_TARGET_GAIN:
- g_value_set_double (value, self->target_gain);
- break;
- case PROP_RESULT_GAIN:
- g_value_set_double (value, self->result_gain);
- break;
- default:
- G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
- break;
- }
-}
-
-static void
-gst_rg_volume_dispose (GObject * object)
-{
- GstRgVolume *self = GST_RG_VOLUME (object);
-
- if (self->volume_element != NULL) {
- /* Manually remove our child using the bin implementation of remove_element.
- * This is needed because we prevent gst_bin_remove from working, which the
- * parent dispose handler would use if we had any children left. */
- GST_BIN_CLASS (parent_class)->remove_element (GST_BIN_CAST (self),
- self->volume_element);
- self->volume_element = NULL;
- }
-
- G_OBJECT_CLASS (parent_class)->dispose (object);
-}
-
-static GstStateChangeReturn
-gst_rg_volume_change_state (GstElement * element, GstStateChange transition)
-{
- GstRgVolume *self = GST_RG_VOLUME (element);
- GstStateChangeReturn res;
-
- switch (transition) {
- case GST_STATE_CHANGE_NULL_TO_READY:
-
- if (G_UNLIKELY (self->volume_element == NULL)) {
- /* Creating our child volume element in _init failed. */
- return GST_STATE_CHANGE_FAILURE;
- }
- break;
-
- case GST_STATE_CHANGE_READY_TO_PAUSED:
-
- gst_rg_volume_reset (self);
- break;
-
- default:
- break;
- }
-
- res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
-
- return res;
-}
-
-/* Event function for the ghost sink pad. */
-static gboolean
-gst_rg_volume_sink_event (GstPad * pad, GstEvent * event)
-{
- GstRgVolume *self;
- GstPad *volume_sink_pad;
- GstEvent *send_event = event;
- gboolean res;
-
- self = GST_RG_VOLUME (gst_pad_get_parent_element (pad));
- volume_sink_pad = gst_ghost_pad_get_target (GST_GHOST_PAD (pad));
-
- switch (GST_EVENT_TYPE (event)) {
- case GST_EVENT_TAG:
-
- GST_LOG_OBJECT (self, "received tag event");
-
- send_event = gst_rg_volume_tag_event (self, event);
-
- if (send_event == NULL)
- GST_LOG_OBJECT (self, "all tags handled, dropping event");
-
- break;
-
- case GST_EVENT_EOS:
-
- gst_rg_volume_reset (self);
- break;
-
- default:
- break;
- }
-
- if (G_LIKELY (send_event != NULL))
- res = gst_pad_send_event (volume_sink_pad, send_event);
- else
- res = TRUE;
-
- gst_object_unref (volume_sink_pad);
- gst_object_unref (self);
- return res;
-}
-
-static GstEvent *
-gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event)
-{
- GstTagList *tag_list;
- gboolean has_track_gain, has_track_peak, has_album_gain, has_album_peak;
- gboolean has_ref_level;
-
- g_return_val_if_fail (event != NULL, NULL);
- g_return_val_if_fail (GST_EVENT_TYPE (event) == GST_EVENT_TAG, event);
-
- gst_event_parse_tag (event, &tag_list);
-
- if (gst_tag_list_is_empty (tag_list))
- return event;
-
- has_track_gain = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN,
- &self->track_gain);
- has_track_peak = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK,
- &self->track_peak);
- has_album_gain = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN,
- &self->album_gain);
- has_album_peak = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK,
- &self->album_peak);
- has_ref_level = gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
- &self->reference_level);
-
- if (!has_track_gain && !has_track_peak && !has_album_gain && !has_album_peak)
- return event;
-
- if (has_ref_level && (has_track_gain || has_album_gain)
- && (ABS (self->reference_level - RG_REFERENCE_LEVEL) > 1.e-6)) {
- /* Log a message stating the amount of adjustment that is applied below. */
- GST_DEBUG_OBJECT (self,
- "compensating for reference level difference by %" GAIN_FORMAT,
- RG_REFERENCE_LEVEL - self->reference_level);
- }
- if (has_track_gain) {
- self->track_gain += RG_REFERENCE_LEVEL - self->reference_level;
- }
- if (has_album_gain) {
- self->album_gain += RG_REFERENCE_LEVEL - self->reference_level;
- }
-
- /* Ignore values that are obviously invalid. */
- if (G_UNLIKELY (has_track_gain && !VALID_GAIN (self->track_gain))) {
- GST_DEBUG_OBJECT (self,
- "ignoring bogus track gain value %" GAIN_FORMAT, self->track_gain);
- has_track_gain = FALSE;
- }
- if (G_UNLIKELY (has_track_peak && !VALID_PEAK (self->track_peak))) {
- GST_DEBUG_OBJECT (self,
- "ignoring bogus track peak value %" PEAK_FORMAT, self->track_peak);
- has_track_peak = FALSE;
- }
- if (G_UNLIKELY (has_album_gain && !VALID_GAIN (self->album_gain))) {
- GST_DEBUG_OBJECT (self,
- "ignoring bogus album gain value %" GAIN_FORMAT, self->album_gain);
- has_album_gain = FALSE;
- }
- if (G_UNLIKELY (has_album_peak && !VALID_PEAK (self->album_peak))) {
- GST_DEBUG_OBJECT (self,
- "ignoring bogus album peak value %" PEAK_FORMAT, self->album_peak);
- has_album_peak = FALSE;
- }
-
- self->has_track_gain |= has_track_gain;
- self->has_track_peak |= has_track_peak;
- self->has_album_gain |= has_album_gain;
- self->has_album_peak |= has_album_peak;
-
- event = (GstEvent *) gst_mini_object_make_writable (GST_MINI_OBJECT (event));
- gst_event_parse_tag (event, &tag_list);
-
- gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_GAIN);
- gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_PEAK);
- gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_GAIN);
- gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_PEAK);
- gst_tag_list_remove_tag (tag_list, GST_TAG_REFERENCE_LEVEL);
-
- gst_rg_volume_update_gain (self);
-
- if (gst_tag_list_is_empty (tag_list)) {
- gst_event_unref (event);
- event = NULL;
- }
-
- return event;
-}
-
-static void
-gst_rg_volume_reset (GstRgVolume * self)
-{
- self->has_track_gain = FALSE;
- self->has_track_peak = FALSE;
- self->has_album_gain = FALSE;
- self->has_album_peak = FALSE;
-
- self->reference_level = RG_REFERENCE_LEVEL;
-
- gst_rg_volume_update_gain (self);
-}
-
-static void
-gst_rg_volume_update_gain (GstRgVolume * self)
-{
- gdouble target_gain, result_gain, result_volume;
- gboolean target_gain_changed, result_gain_changed;
-
- gst_rg_volume_determine_gain (self, &target_gain, &result_gain);
-
- result_volume = DB_TO_LINEAR (result_gain);
-
- /* Ensure that the result volume is within the range that the volume element
- * can handle. Currently, the limit is 10. (+20 dB), which should not be
- * restrictive. */
- if (G_UNLIKELY (result_volume > self->max_volume)) {
- GST_INFO_OBJECT (self,
- "cannot handle result gain of %" GAIN_FORMAT " (%0.6f), adjusting",
- result_gain, result_volume);
-
- result_volume = self->max_volume;
- result_gain = LINEAR_TO_DB (result_volume);
- }
-
- /* Direct comparison is OK in this case. */
- if (target_gain == result_gain) {
- GST_INFO_OBJECT (self,
- "result gain is %" GAIN_FORMAT " (%0.6f), matching target",
- result_gain, result_volume);
- } else {
- GST_INFO_OBJECT (self,
- "result gain is %" GAIN_FORMAT " (%0.6f), target is %" GAIN_FORMAT,
- result_gain, result_volume, target_gain);
- }
-
- target_gain_changed = (self->target_gain != target_gain);
- result_gain_changed = (self->result_gain != result_gain);
-
- self->target_gain = target_gain;
- self->result_gain = result_gain;
-
- g_object_set (self->volume_element, "volume", result_volume, NULL);
-
- if (target_gain_changed)
- g_object_notify ((GObject *) self, "target-gain");
- if (result_gain_changed)
- g_object_notify ((GObject *) self, "result-gain");
-}
-
-static inline void
-gst_rg_volume_determine_gain (GstRgVolume * self, gdouble * target_gain,
- gdouble * result_gain)
-{
- gdouble gain, peak;
-
- if (!self->has_track_gain && !self->has_album_gain) {
-
- GST_DEBUG_OBJECT (self, "using fallback gain");
- gain = self->fallback_gain;
- peak = 1.0;
-
- } else if ((self->album_mode && self->has_album_gain)
- || (!self->album_mode && !self->has_track_gain)) {
-
- gain = self->album_gain;
- if (G_LIKELY (self->has_album_peak)) {
- peak = self->album_peak;
- } else {
- GST_DEBUG_OBJECT (self, "album peak missing, assuming 1.0");
- peak = 1.0;
- }
- /* Falling back from track to album gain shouldn't really happen. */
- if (G_UNLIKELY (!self->album_mode))
- GST_INFO_OBJECT (self, "falling back to album gain");
-
- } else {
- /* !album_mode && !has_album_gain || album_mode && has_track_gain */
-
- gain = self->track_gain;
- if (G_LIKELY (self->has_track_peak)) {
- peak = self->track_peak;
- } else {
- GST_DEBUG_OBJECT (self, "track peak missing, assuming 1.0");
- peak = 1.0;
- }
- if (self->album_mode)
- GST_INFO_OBJECT (self, "falling back to track gain");
- }
-
- gain += self->pre_amp;
-
- *target_gain = gain;
- *result_gain = gain;
-
- if (LINEAR_TO_DB (peak) + gain > self->headroom) {
- *result_gain = LINEAR_TO_DB (1. / peak) + self->headroom;
- }
-}
diff --git a/gst/replaygain/gstrgvolume.h b/gst/replaygain/gstrgvolume.h
deleted file mode 100644
index a0a5a8ce..00000000
--- a/gst/replaygain/gstrgvolume.h
+++ /dev/null
@@ -1,88 +0,0 @@
-/* GStreamer ReplayGain volume adjustment
- *
- * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
- *
- * gstrgvolume.h: Element to apply ReplayGain volume adjustment
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-#ifndef __GST_RG_VOLUME_H__
-#define __GST_RG_VOLUME_H__
-
-#include <gst/gst.h>
-
-G_BEGIN_DECLS
-
-#define GST_TYPE_RG_VOLUME \
- (gst_rg_volume_get_type())
-#define GST_RG_VOLUME(obj) \
- (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_VOLUME,GstRgVolume))
-#define GST_RG_VOLUME_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_VOLUME,GstRgVolumeClass))
-#define GST_IS_RG_VOLUME(obj) \
- (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_VOLUME))
-#define GST_IS_RG_VOLUME_CLASS(klass) \
- (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_VOLUME))
-
-typedef struct _GstRgVolume GstRgVolume;
-typedef struct _GstRgVolumeClass GstRgVolumeClass;
-
-/**
- * GstRgVolume:
- *
- * Opaque data structure.
- */
-struct _GstRgVolume
-{
- GstBin bin;
-
- /*< private >*/
-
- GstElement *volume_element;
- gdouble max_volume;
-
- gboolean album_mode;
- gdouble headroom;
- gdouble pre_amp;
- gdouble fallback_gain;
-
- gdouble target_gain;
- gdouble result_gain;
-
- gdouble track_gain;
- gdouble track_peak;
- gdouble album_gain;
- gdouble album_peak;
-
- gboolean has_track_gain;
- gboolean has_track_peak;
- gboolean has_album_gain;
- gboolean has_album_peak;
-
- gdouble reference_level;
-};
-
-struct _GstRgVolumeClass
-{
- GstBinClass parent_class;
-};
-
-GType gst_rg_volume_get_type (void);
-
-G_END_DECLS
-
-#endif /* __GST_RG_VOLUME_H__ */
diff --git a/gst/replaygain/replaygain.c b/gst/replaygain/replaygain.c
deleted file mode 100644
index d0127e8b..00000000
--- a/gst/replaygain/replaygain.c
+++ /dev/null
@@ -1,53 +0,0 @@
-/* GStreamer ReplayGain plugin
- *
- * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
- *
- * replaygain.c: Plugin providing ReplayGain related elements
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#include <gst/gst.h>
-
-#include "gstrganalysis.h"
-#include "gstrglimiter.h"
-#include "gstrgvolume.h"
-
-static gboolean
-plugin_init (GstPlugin * plugin)
-{
- if (!gst_element_register (plugin, "rganalysis", GST_RANK_NONE,
- GST_TYPE_RG_ANALYSIS))
- return FALSE;
-
- if (!gst_element_register (plugin, "rglimiter", GST_RANK_NONE,
- GST_TYPE_RG_LIMITER))
- return FALSE;
-
- if (!gst_element_register (plugin, "rgvolume", GST_RANK_NONE,
- GST_TYPE_RG_VOLUME))
- return FALSE;
-
- return TRUE;
-}
-
-GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "replaygain",
- "ReplayGain volume normalization", plugin_init, VERSION, GST_LICENSE,
- GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN);
diff --git a/gst/replaygain/replaygain.h b/gst/replaygain/replaygain.h
deleted file mode 100644
index 15be8885..00000000
--- a/gst/replaygain/replaygain.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/* GStreamer ReplayGain plugin
- *
- * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
- *
- * replaygain.h: Plugin providing ReplayGain related elements
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-#ifndef __REPLAYGAIN_H__
-#define __REPLAYGAIN_H__
-
-G_BEGIN_DECLS
-
-/* Reference level (in dBSPL). The 2001 proposal specifies 83. This was
- * changed later in all implementations to 89, which is the new, offical value:
- * David Robinson acknowledged the change but didn't update the website yet. */
-
-#define RG_REFERENCE_LEVEL 89.
-
-G_END_DECLS
-
-#endif /* __REPLAYGAIN_H__ */
diff --git a/gst/replaygain/rganalysis.c b/gst/replaygain/rganalysis.c
deleted file mode 100644
index 147eef85..00000000
--- a/gst/replaygain/rganalysis.c
+++ /dev/null
@@ -1,777 +0,0 @@
-/* GStreamer ReplayGain analysis
- *
- * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
- * Copyright (C) 2001 David Robinson <David@Robinson.org>
- * Glen Sawyer <glensawyer@hotmail.com>
- *
- * rganalysis.c: Analyze raw audio data in accordance with ReplayGain
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-/* Based on code with Copyright (C) 2001 David Robinson
- * <David@Robinson.org> and Glen Sawyer <glensawyer@hotmail.com>,
- * which is distributed under the LGPL as part of the vorbisgain
- * program. The original code also mentions Frank Klemm
- * (http://www.uni-jena.de/~pfk/mpp/) for having contributed lots of
- * good code. Specifically, this is based on the file
- * "gain_analysis.c" from vorbisgain version 0.34.
- */
-
-/* Room for future improvement: Mono data is currently in fact copied
- * to two channels which get processed normally. This means that mono
- * input data is processed twice.
- */
-
-/* Helpful information for understanding this code: The two IIR
- * filters depend on previous input _and_ previous output samples (up
- * to the filter's order number of samples). This explains the whole
- * lot of memcpy'ing done in rg_analysis_analyze and why the context
- * holds so many buffers.
- */
-
-#include <math.h>
-#include <string.h>
-#include <glib.h>
-
-#include "rganalysis.h"
-
-#define YULE_ORDER 10
-#define BUTTER_ORDER 2
-/* Percentile which is louder than the proposed level: */
-#define RMS_PERCENTILE 95
-/* Duration of RMS window in milliseconds: */
-#define RMS_WINDOW_MSECS 50
-/* Histogram array elements per dB: */
-#define STEPS_PER_DB 100
-/* Histogram upper bound in dB (normal max. values in the wild are
- * assumed to be around 70, 80 dB): */
-#define MAX_DB 120
-/* Calibration value: */
-#define PINK_REF 64.82 /* 298640883795 */
-
-#define MAX_ORDER MAX (BUTTER_ORDER, YULE_ORDER)
-#define MAX_SAMPLE_RATE 48000
-/* The + 999 has the effect of ceil()ing: */
-#define MAX_SAMPLE_WINDOW (guint) \
- ((MAX_SAMPLE_RATE * RMS_WINDOW_MSECS + 999) / 1000)
-
-/* Analysis result accumulator. */
-
-struct _RgAnalysisAcc
-{
- guint32 histogram[STEPS_PER_DB * MAX_DB];
- gdouble peak;
-};
-
-typedef struct _RgAnalysisAcc RgAnalysisAcc;
-
-/* Analysis context. */
-
-struct _RgAnalysisCtx
-{
- /* Filter buffers for left channel. */
- gfloat inprebuf_l[MAX_ORDER * 2];
- gfloat *inpre_l;
- gfloat stepbuf_l[MAX_SAMPLE_WINDOW + MAX_ORDER];
- gfloat *step_l;
- gfloat outbuf_l[MAX_SAMPLE_WINDOW + MAX_ORDER];
- gfloat *out_l;
- /* Filter buffers for right channel. */
- gfloat inprebuf_r[MAX_ORDER * 2];
- gfloat *inpre_r;
- gfloat stepbuf_r[MAX_SAMPLE_WINDOW + MAX_ORDER];
- gfloat *step_r;
- gfloat outbuf_r[MAX_SAMPLE_WINDOW + MAX_ORDER];
- gfloat *out_r;
-
- /* Number of samples to reach duration of the RMS window: */
- guint window_n_samples;
- /* Progress of the running window: */
- guint window_n_samples_done;
- gdouble window_square_sum;
-
- gint sample_rate;
- gint sample_rate_index;
-
- RgAnalysisAcc track;
- RgAnalysisAcc album;
-};
-
-/* Filter coefficients for the IIR filters that form the equal
- * loudness filter. XFilter[ctx->sample_rate_index] gives the array
- * of the X coefficients (A or B) for the configured sample rate. */
-
-#ifdef _MSC_VER
-/* Disable double-to-float warning: */
-/* A better solution would be to append 'f' to each constant, but that
- * makes the code ugly. */
-#pragma warning ( disable : 4305 )
-#endif
-
-static const gfloat AYule[9][11] = {
- {1., -3.84664617118067, 7.81501653005538, -11.34170355132042,
- 13.05504219327545, -12.28759895145294, 9.48293806319790,
- -5.87257861775999, 2.75465861874613, -0.86984376593551,
- 0.13919314567432},
- {1., -3.47845948550071, 6.36317777566148, -8.54751527471874, 9.47693607801280,
- -8.81498681370155, 6.85401540936998, -4.39470996079559,
- 2.19611684890774, -0.75104302451432, 0.13149317958808},
- {1., -2.37898834973084, 2.84868151156327, -2.64577170229825, 2.23697657451713,
- -1.67148153367602, 1.00595954808547, -0.45953458054983,
- 0.16378164858596, -0.05032077717131, 0.02347897407020},
- {1., -1.61273165137247, 1.07977492259970, -0.25656257754070,
- -0.16276719120440, -0.22638893773906, 0.39120800788284,
- -0.22138138954925, 0.04500235387352, 0.02005851806501,
- 0.00302439095741},
- {1., -1.49858979367799, 0.87350271418188, 0.12205022308084, -0.80774944671438,
- 0.47854794562326, -0.12453458140019, -0.04067510197014,
- 0.08333755284107, -0.04237348025746, 0.02977207319925},
- {1., -0.62820619233671, 0.29661783706366, -0.37256372942400, 0.00213767857124,
- -0.42029820170918, 0.22199650564824, 0.00613424350682, 0.06747620744683,
- 0.05784820375801, 0.03222754072173},
- {1., -1.04800335126349, 0.29156311971249, -0.26806001042947, 0.00819999645858,
- 0.45054734505008, -0.33032403314006, 0.06739368333110,
- -0.04784254229033, 0.01639907836189, 0.01807364323573},
- {1., -0.51035327095184, -0.31863563325245, -0.20256413484477,
- 0.14728154134330, 0.38952639978999, -0.23313271880868,
- -0.05246019024463, -0.02505961724053, 0.02442357316099,
- 0.01818801111503},
- {1., -0.25049871956020, -0.43193942311114, -0.03424681017675,
- -0.04678328784242, 0.26408300200955, 0.15113130533216,
- -0.17556493366449, -0.18823009262115, 0.05477720428674,
- 0.04704409688120}
-};
-
-static const gfloat BYule[9][11] = {
- {0.03857599435200, -0.02160367184185, -0.00123395316851, -0.00009291677959,
- -0.01655260341619, 0.02161526843274, -0.02074045215285,
- 0.00594298065125, 0.00306428023191, 0.00012025322027, 0.00288463683916},
- {0.05418656406430, -0.02911007808948, -0.00848709379851, -0.00851165645469,
- -0.00834990904936, 0.02245293253339, -0.02596338512915,
- 0.01624864962975, -0.00240879051584, 0.00674613682247,
- -0.00187763777362},
- {0.15457299681924, -0.09331049056315, -0.06247880153653, 0.02163541888798,
- -0.05588393329856, 0.04781476674921, 0.00222312597743, 0.03174092540049,
- -0.01390589421898, 0.00651420667831, -0.00881362733839},
- {0.30296907319327, -0.22613988682123, -0.08587323730772, 0.03282930172664,
- -0.00915702933434, -0.02364141202522, -0.00584456039913,
- 0.06276101321749, -0.00000828086748, 0.00205861885564,
- -0.02950134983287},
- {0.33642304856132, -0.25572241425570, -0.11828570177555, 0.11921148675203,
- -0.07834489609479, -0.00469977914380, -0.00589500224440,
- 0.05724228140351, 0.00832043980773, -0.01635381384540,
- -0.01760176568150},
- {0.44915256608450, -0.14351757464547, -0.22784394429749, -0.01419140100551,
- 0.04078262797139, -0.12398163381748, 0.04097565135648, 0.10478503600251,
- -0.01863887810927, -0.03193428438915, 0.00541907748707},
- {0.56619470757641, -0.75464456939302, 0.16242137742230, 0.16744243493672,
- -0.18901604199609, 0.30931782841830, -0.27562961986224,
- 0.00647310677246, 0.08647503780351, -0.03788984554840,
- -0.00588215443421},
- {0.58100494960553, -0.53174909058578, -0.14289799034253, 0.17520704835522,
- 0.02377945217615, 0.15558449135573, -0.25344790059353, 0.01628462406333,
- 0.06920467763959, -0.03721611395801, -0.00749618797172},
- {0.53648789255105, -0.42163034350696, -0.00275953611929, 0.04267842219415,
- -0.10214864179676, 0.14590772289388, -0.02459864859345,
- -0.11202315195388, -0.04060034127000, 0.04788665548180,
- -0.02217936801134}
-};
-
-static const gfloat AButter[9][3] = {
- {1., -1.97223372919527, 0.97261396931306},
- {1., -1.96977855582618, 0.97022847566350},
- {1., -1.95835380975398, 0.95920349965459},
- {1., -1.95002759149878, 0.95124613669835},
- {1., -1.94561023566527, 0.94705070426118},
- {1., -1.92783286977036, 0.93034775234268},
- {1., -1.91858953033784, 0.92177618768381},
- {1., -1.91542108074780, 0.91885558323625},
- {1., -1.88903307939452, 0.89487434461664}
-};
-
-static const gfloat BButter[9][3] = {
- {0.98621192462708, -1.97242384925416, 0.98621192462708},
- {0.98500175787242, -1.97000351574484, 0.98500175787242},
- {0.97938932735214, -1.95877865470428, 0.97938932735214},
- {0.97531843204928, -1.95063686409857, 0.97531843204928},
- {0.97316523498161, -1.94633046996323, 0.97316523498161},
- {0.96454515552826, -1.92909031105652, 0.96454515552826},
- {0.96009142950541, -1.92018285901082, 0.96009142950541},
- {0.95856916599601, -1.91713833199203, 0.95856916599601},
- {0.94597685600279, -1.89195371200558, 0.94597685600279}
-};
-
-#ifdef _MSC_VER
-#pragma warning ( default : 4305 )
-#endif
-
-/* Filter functions. These access elements with negative indices of
- * the input and output arrays (up to the filter's order). */
-
-/* For much better performance, the function below has been
- * implemented by unrolling the inner loop for our two use cases. */
-
-/*
- * static inline void
- * apply_filter (const gfloat * input, gfloat * output, guint n_samples,
- * const gfloat * a, const gfloat * b, guint order)
- * {
- * gfloat y;
- * gint i, k;
- *
- * for (i = 0; i < n_samples; i++) {
- * y = input[i] * b[0];
- * for (k = 1; k <= order; k++)
- * y += input[i - k] * b[k] - output[i - k] * a[k];
- * output[i] = y;
- * }
- * }
- */
-
-static inline void
-yule_filter (const gfloat * input, gfloat * output,
- const gfloat * a, const gfloat * b)
-{
- /* 1e-10 is added below to avoid running into denormals when operating on
- * near silence. */
-
- output[0] = 1e-10 + input[0] * b[0]
- + input[-1] * b[1] - output[-1] * a[1]
- + input[-2] * b[2] - output[-2] * a[2]
- + input[-3] * b[3] - output[-3] * a[3]
- + input[-4] * b[4] - output[-4] * a[4]
- + input[-5] * b[5] - output[-5] * a[5]
- + input[-6] * b[6] - output[-6] * a[6]
- + input[-7] * b[7] - output[-7] * a[7]
- + input[-8] * b[8] - output[-8] * a[8]
- + input[-9] * b[9] - output[-9] * a[9]
- + input[-10] * b[10] - output[-10] * a[10];
-}
-
-static inline void
-butter_filter (const gfloat * input, gfloat * output,
- const gfloat * a, const gfloat * b)
-{
- output[0] = input[0] * b[0]
- + input[-1] * b[1] - output[-1] * a[1]
- + input[-2] * b[2] - output[-2] * a[2];
-}
-
-/* Because butter_filter and yule_filter are inlined, this function is
- * a bit blown-up (code-size wise), but not inlining gives a ca. 40%
- * performance penalty. */
-
-static inline void
-apply_filters (const RgAnalysisCtx * ctx, const gfloat * input_l,
- const gfloat * input_r, guint n_samples)
-{
- const gfloat *ayule = AYule[ctx->sample_rate_index];
- const gfloat *byule = BYule[ctx->sample_rate_index];
- const gfloat *abutter = AButter[ctx->sample_rate_index];
- const gfloat *bbutter = BButter[ctx->sample_rate_index];
- gint pos = ctx->window_n_samples_done;
- gint i;
-
- for (i = 0; i < n_samples; i++, pos++) {
- yule_filter (input_l + i, ctx->step_l + pos, ayule, byule);
- butter_filter (ctx->step_l + pos, ctx->out_l + pos, abutter, bbutter);
-
- yule_filter (input_r + i, ctx->step_r + pos, ayule, byule);
- butter_filter (ctx->step_r + pos, ctx->out_r + pos, abutter, bbutter);
- }
-}
-
-/* Clear filter buffer state and current RMS window. */
-
-static void
-reset_filters (RgAnalysisCtx * ctx)
-{
- gint i;
-
- for (i = 0; i < MAX_ORDER; i++) {
-
- ctx->inprebuf_l[i] = 0.;
- ctx->stepbuf_l[i] = 0.;
- ctx->outbuf_l[i] = 0.;
-
- ctx->inprebuf_r[i] = 0.;
- ctx->stepbuf_r[i] = 0.;
- ctx->outbuf_r[i] = 0.;
- }
-
- ctx->window_square_sum = 0.;
- ctx->window_n_samples_done = 0;
-}
-
-/* Accumulator functions. */
-
-/* Add two accumulators in-place. The sum is defined as the result of
- * the vector sum of the histogram array and the maximum value of the
- * peak field. Thus "adding" the accumulators for all tracks yields
- * the correct result for obtaining the album gain and peak. */
-
-static void
-accumulator_add (RgAnalysisAcc * acc, const RgAnalysisAcc * acc_other)
-{
- gint i;
-
- for (i = 0; i < G_N_ELEMENTS (acc->histogram); i++)
- acc->histogram[i] += acc_other->histogram[i];
-
- acc->peak = MAX (acc->peak, acc_other->peak);
-}
-
-/* Reset an accumulator to zero. */
-
-static void
-accumulator_clear (RgAnalysisAcc * acc)
-{
- memset (acc->histogram, 0, sizeof (acc->histogram));
- acc->peak = 0.;
-}
-
-/* Obtain final analysis result from an accumulator. Returns TRUE on
- * success, FALSE on error (if accumulator is still zero). */
-
-static gboolean
-accumulator_result (const RgAnalysisAcc * acc, gdouble * result_gain,
- gdouble * result_peak)
-{
- guint32 sum = 0;
- guint32 upper;
- guint i;
-
- for (i = 0; i < G_N_ELEMENTS (acc->histogram); i++)
- sum += acc->histogram[i];
-
- if (sum == 0)
- /* All entries are 0: We got less than 50ms of data. */
- return FALSE;
-
- upper = (guint32) ceil (sum * (1. - (gdouble) (RMS_PERCENTILE / 100.)));
-
- for (i = G_N_ELEMENTS (acc->histogram); i--;) {
- if (upper <= acc->histogram[i])
- break;
- upper -= acc->histogram[i];
- }
-
- if (result_peak != NULL)
- *result_peak = acc->peak;
- if (result_gain != NULL)
- *result_gain = PINK_REF - (gdouble) i / STEPS_PER_DB;
-
- return TRUE;
-}
-
-/* Functions that operate on contexts, for external usage. */
-
-/* Create a new context. Before it can be used, a sample rate must be
- * configured using rg_analysis_set_sample_rate. */
-
-RgAnalysisCtx *
-rg_analysis_new (void)
-{
- RgAnalysisCtx *ctx;
-
- ctx = g_new (RgAnalysisCtx, 1);
-
- ctx->inpre_l = ctx->inprebuf_l + MAX_ORDER;
- ctx->step_l = ctx->stepbuf_l + MAX_ORDER;
- ctx->out_l = ctx->outbuf_l + MAX_ORDER;
-
- ctx->inpre_r = ctx->inprebuf_r + MAX_ORDER;
- ctx->step_r = ctx->stepbuf_r + MAX_ORDER;
- ctx->out_r = ctx->outbuf_r + MAX_ORDER;
-
- ctx->sample_rate = 0;
-
- accumulator_clear (&ctx->track);
- accumulator_clear (&ctx->album);
-
- return ctx;
-}
-
-/* Adapt to given sample rate. Does nothing if already the current
- * rate (returns TRUE then). Returns FALSE only if given sample rate
- * is not supported. If the configured rate changes, the last
- * unprocessed incomplete 50ms chunk of data is dropped because the
- * filters are reset. */
-
-gboolean
-rg_analysis_set_sample_rate (RgAnalysisCtx * ctx, gint sample_rate)
-{
- g_return_val_if_fail (ctx != NULL, FALSE);
-
- if (ctx->sample_rate == sample_rate)
- return TRUE;
-
- switch (sample_rate) {
- case 48000:
- ctx->sample_rate_index = 0;
- break;
- case 44100:
- ctx->sample_rate_index = 1;
- break;
- case 32000:
- ctx->sample_rate_index = 2;
- break;
- case 24000:
- ctx->sample_rate_index = 3;
- break;
- case 22050:
- ctx->sample_rate_index = 4;
- break;
- case 16000:
- ctx->sample_rate_index = 5;
- break;
- case 12000:
- ctx->sample_rate_index = 6;
- break;
- case 11025:
- ctx->sample_rate_index = 7;
- break;
- case 8000:
- ctx->sample_rate_index = 8;
- break;
- default:
- return FALSE;
- }
-
- ctx->sample_rate = sample_rate;
- /* The + 999 has the effect of ceil()ing: */
- ctx->window_n_samples = (guint) ((sample_rate * RMS_WINDOW_MSECS + 999)
- / 1000);
-
- reset_filters (ctx);
-
- return TRUE;
-}
-
-void
-rg_analysis_destroy (RgAnalysisCtx * ctx)
-{
- g_free (ctx);
-}
-
-/* Entry points for analyzing sample data in common raw data formats.
- * The stereo format functions expect interleaved frames. It is
- * possible to pass data in different formats for the same context,
- * there are no restrictions. All functions have the same signature;
- * the depth argument for the float functions is not variable and must
- * be given the value 32. */
-
-void
-rg_analysis_analyze_mono_float (RgAnalysisCtx * ctx, gconstpointer data,
- gsize size, guint depth)
-{
- gfloat conv_samples[512];
- const gfloat *samples = (gfloat *) data;
- guint n_samples = size / sizeof (gfloat);
- gint i;
-
- g_return_if_fail (depth == 32);
- g_return_if_fail (size % sizeof (gfloat) == 0);
-
- while (n_samples) {
- gint n = MIN (n_samples, G_N_ELEMENTS (conv_samples));
-
- n_samples -= n;
- memcpy (conv_samples, samples, n * sizeof (gfloat));
- for (i = 0; i < n; i++) {
- ctx->track.peak = MAX (ctx->track.peak, fabs (conv_samples[i]));
- conv_samples[i] *= 32768.;
- }
- samples += n;
- rg_analysis_analyze (ctx, conv_samples, NULL, n);
- }
-}
-
-void
-rg_analysis_analyze_stereo_float (RgAnalysisCtx * ctx, gconstpointer data,
- gsize size, guint depth)
-{
- gfloat conv_samples_l[256];
- gfloat conv_samples_r[256];
- const gfloat *samples = (gfloat *) data;
- guint n_frames = size / (sizeof (gfloat) * 2);
- gint i;
-
- g_return_if_fail (depth == 32);
- g_return_if_fail (size % (sizeof (gfloat) * 2) == 0);
-
- while (n_frames) {
- gint n = MIN (n_frames, G_N_ELEMENTS (conv_samples_l));
-
- n_frames -= n;
- for (i = 0; i < n; i++) {
- gfloat old_sample;
-
- old_sample = samples[2 * i];
- ctx->track.peak = MAX (ctx->track.peak, fabs (old_sample));
- conv_samples_l[i] = old_sample * 32768.;
-
- old_sample = samples[2 * i + 1];
- ctx->track.peak = MAX (ctx->track.peak, fabs (old_sample));
- conv_samples_r[i] = old_sample * 32768.;
- }
- samples += 2 * n;
- rg_analysis_analyze (ctx, conv_samples_l, conv_samples_r, n);
- }
-}
-
-void
-rg_analysis_analyze_mono_int16 (RgAnalysisCtx * ctx, gconstpointer data,
- gsize size, guint depth)
-{
- gfloat conv_samples[512];
- gint32 peak_sample = 0;
- const gint16 *samples = (gint16 *) data;
- guint n_samples = size / sizeof (gint16);
- gint shift = sizeof (gint16) * 8 - depth;
- gint i;
-
- g_return_if_fail (depth <= (sizeof (gint16) * 8));
- g_return_if_fail (size % sizeof (gint16) == 0);
-
- while (n_samples) {
- gint n = MIN (n_samples, G_N_ELEMENTS (conv_samples));
-
- n_samples -= n;
- for (i = 0; i < n; i++) {
- gint16 old_sample = samples[i] << shift;
-
- peak_sample = MAX (peak_sample, ABS ((gint32) old_sample));
- conv_samples[i] = (gfloat) old_sample;
- }
- samples += n;
- rg_analysis_analyze (ctx, conv_samples, NULL, n);
- }
- ctx->track.peak = MAX (ctx->track.peak,
- (gdouble) peak_sample / ((gdouble) (1u << 15)));
-}
-
-void
-rg_analysis_analyze_stereo_int16 (RgAnalysisCtx * ctx, gconstpointer data,
- gsize size, guint depth)
-{
- gfloat conv_samples_l[256];
- gfloat conv_samples_r[256];
- gint32 peak_sample = 0;
- const gint16 *samples = (gint16 *) data;
- guint n_frames = size / (sizeof (gint16) * 2);
- gint shift = sizeof (gint16) * 8 - depth;
- gint i;
-
- g_return_if_fail (depth <= (sizeof (gint16) * 8));
- g_return_if_fail (size % (sizeof (gint16) * 2) == 0);
-
- while (n_frames) {
- gint n = MIN (n_frames, G_N_ELEMENTS (conv_samples_l));
-
- n_frames -= n;
- for (i = 0; i < n; i++) {
- gint16 old_sample;
-
- old_sample = samples[2 * i] << shift;
- peak_sample = MAX (peak_sample, ABS ((gint32) old_sample));
- conv_samples_l[i] = (gfloat) old_sample;
-
- old_sample = samples[2 * i + 1] << shift;
- peak_sample = MAX (peak_sample, ABS ((gint32) old_sample));
- conv_samples_r[i] = (gfloat) old_sample;
- }
- samples += 2 * n;
- rg_analysis_analyze (ctx, conv_samples_l, conv_samples_r, n);
- }
- ctx->track.peak = MAX (ctx->track.peak,
- (gdouble) peak_sample / ((gdouble) (1u << 15)));
-}
-
-/* Analyze the given chunk of samples. The sample data is given in
- * floating point format but should be scaled such that the values
- * +/-32768.0 correspond to the -0dBFS reference amplitude.
- *
- * samples_l: Buffer with sample data for the left channel or of the
- * mono channel.
- *
- * samples_r: Buffer with sample data for the right channel or NULL
- * for mono.
- *
- * n_samples: Number of samples passed in each buffer.
- */
-
-void
-rg_analysis_analyze (RgAnalysisCtx * ctx, const gfloat * samples_l,
- const gfloat * samples_r, guint n_samples)
-{
- const gfloat *input_l, *input_r;
- guint n_samples_done;
- gint i;
-
- g_return_if_fail (ctx != NULL);
- g_return_if_fail (samples_l != NULL);
- g_return_if_fail (ctx->sample_rate != 0);
-
- if (n_samples == 0)
- return;
-
- if (samples_r == NULL)
- /* Mono. */
- samples_r = samples_l;
-
- memcpy (ctx->inpre_l, samples_l,
- MIN (n_samples, MAX_ORDER) * sizeof (gfloat));
- memcpy (ctx->inpre_r, samples_r,
- MIN (n_samples, MAX_ORDER) * sizeof (gfloat));
-
- n_samples_done = 0;
- while (n_samples_done < n_samples) {
- /* Limit number of samples to be processed in this iteration to
- * the number needed to complete the next window: */
- guint n_samples_current = MIN (n_samples - n_samples_done,
- ctx->window_n_samples - ctx->window_n_samples_done);
-
- if (n_samples_done < MAX_ORDER) {
- input_l = ctx->inpre_l + n_samples_done;
- input_r = ctx->inpre_r + n_samples_done;
- n_samples_current = MIN (n_samples_current, MAX_ORDER - n_samples_done);
- } else {
- input_l = samples_l + n_samples_done;
- input_r = samples_r + n_samples_done;
- }
-
- apply_filters (ctx, input_l, input_r, n_samples_current);
-
- /* Update the square sum. */
- for (i = 0; i < n_samples_current; i++)
- ctx->window_square_sum += ctx->out_l[ctx->window_n_samples_done + i]
- * ctx->out_l[ctx->window_n_samples_done + i]
- + ctx->out_r[ctx->window_n_samples_done + i]
- * ctx->out_r[ctx->window_n_samples_done + i];
-
- ctx->window_n_samples_done += n_samples_current;
-
- g_return_if_fail (ctx->window_n_samples_done <= ctx->window_n_samples);
-
- if (ctx->window_n_samples_done == ctx->window_n_samples) {
- /* Get the Root Mean Square (RMS) for this set of samples. */
- gdouble val = STEPS_PER_DB * 10. * log10 (ctx->window_square_sum /
- ctx->window_n_samples * 0.5 + 1.e-37);
- gint ival = CLAMP ((gint) val, 0,
- (gint) G_N_ELEMENTS (ctx->track.histogram) - 1);
-
- ctx->track.histogram[ival]++;
- ctx->window_square_sum = 0.;
- ctx->window_n_samples_done = 0;
-
- /* No need for memmove here, the areas never overlap: Even for
- * the smallest sample rate, the number of samples needed for
- * the window is greater than MAX_ORDER. */
-
- memcpy (ctx->stepbuf_l, ctx->stepbuf_l + ctx->window_n_samples,
- MAX_ORDER * sizeof (gfloat));
- memcpy (ctx->outbuf_l, ctx->outbuf_l + ctx->window_n_samples,
- MAX_ORDER * sizeof (gfloat));
-
- memcpy (ctx->stepbuf_r, ctx->stepbuf_r + ctx->window_n_samples,
- MAX_ORDER * sizeof (gfloat));
- memcpy (ctx->outbuf_r, ctx->outbuf_r + ctx->window_n_samples,
- MAX_ORDER * sizeof (gfloat));
- }
-
- n_samples_done += n_samples_current;
- }
-
- if (n_samples >= MAX_ORDER) {
-
- memcpy (ctx->inprebuf_l, samples_l + n_samples - MAX_ORDER,
- MAX_ORDER * sizeof (gfloat));
-
- memcpy (ctx->inprebuf_r, samples_r + n_samples - MAX_ORDER,
- MAX_ORDER * sizeof (gfloat));
-
- } else {
-
- memmove (ctx->inprebuf_l, ctx->inprebuf_l + n_samples,
- (MAX_ORDER - n_samples) * sizeof (gfloat));
- memcpy (ctx->inprebuf_l + MAX_ORDER - n_samples, samples_l,
- n_samples * sizeof (gfloat));
-
- memmove (ctx->inprebuf_r, ctx->inprebuf_r + n_samples,
- (MAX_ORDER - n_samples) * sizeof (gfloat));
- memcpy (ctx->inprebuf_r + MAX_ORDER - n_samples, samples_r,
- n_samples * sizeof (gfloat));
-
- }
-}
-
-/* Obtain track gain and peak. Returns TRUE on success. Can fail if
- * not enough samples have been processed. Updates album accumulator.
- * Resets track accumulator. */
-
-gboolean
-rg_analysis_track_result (RgAnalysisCtx * ctx, gdouble * gain, gdouble * peak)
-{
- gboolean result;
-
- g_return_val_if_fail (ctx != NULL, FALSE);
-
- accumulator_add (&ctx->album, &ctx->track);
- result = accumulator_result (&ctx->track, gain, peak);
- accumulator_clear (&ctx->track);
-
- reset_filters (ctx);
-
- return result;
-}
-
-/* Obtain album gain and peak. Returns TRUE on success. Can fail if
- * not enough samples have been processed. Resets album
- * accumulator. */
-
-gboolean
-rg_analysis_album_result (RgAnalysisCtx * ctx, gdouble * gain, gdouble * peak)
-{
- gboolean result;
-
- g_return_val_if_fail (ctx != NULL, FALSE);
-
- result = accumulator_result (&ctx->album, gain, peak);
- accumulator_clear (&ctx->album);
-
- return result;
-}
-
-void
-rg_analysis_reset_album (RgAnalysisCtx * ctx)
-{
- accumulator_clear (&ctx->album);
-}
-
-/* Reset internal buffers as well as track and album accumulators.
- * Configured sample rate is kept intact. */
-
-void
-rg_analysis_reset (RgAnalysisCtx * ctx)
-{
- g_return_if_fail (ctx != NULL);
-
- reset_filters (ctx);
- accumulator_clear (&ctx->track);
- accumulator_clear (&ctx->album);
-}
diff --git a/gst/replaygain/rganalysis.h b/gst/replaygain/rganalysis.h
deleted file mode 100644
index 16247361..00000000
--- a/gst/replaygain/rganalysis.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/* GStreamer ReplayGain analysis
- *
- * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
- * Copyright (C) 2001 David Robinson <David@Robinson.org>
- * Glen Sawyer <glensawyer@hotmail.com>
- *
- * rganalysis.h: Analyze raw audio data in accordance with ReplayGain
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-#ifndef __RG_ANALYSIS_H__
-#define __RG_ANALYSIS_H__
-
-#include <glib.h>
-
-G_BEGIN_DECLS
-
-typedef struct _RgAnalysisCtx RgAnalysisCtx;
-
-RgAnalysisCtx *rg_analysis_new (void);
-gboolean rg_analysis_set_sample_rate (RgAnalysisCtx * ctx, gint sample_rate);
-void rg_analysis_analyze_mono_float (RgAnalysisCtx * ctx, gconstpointer data,
- gsize size, guint depth);
-void rg_analysis_analyze_stereo_float (RgAnalysisCtx * ctx, gconstpointer data,
- gsize size, guint depth);
-void rg_analysis_analyze_mono_int16 (RgAnalysisCtx * ctx, gconstpointer data,
- gsize size, guint depth);
-void rg_analysis_analyze_stereo_int16 (RgAnalysisCtx * ctx, gconstpointer data,
- gsize size, guint depth);
-void rg_analysis_analyze (RgAnalysisCtx * ctx, const gfloat * samples_l,
- const gfloat * samples_r, guint n_samples);
-gboolean rg_analysis_track_result (RgAnalysisCtx * ctx, gdouble * gain,
- gdouble * peak);
-gboolean rg_analysis_album_result (RgAnalysisCtx * ctx, gdouble * gain,
- gdouble * peak);
-void rg_analysis_reset_album (RgAnalysisCtx * ctx);
-void rg_analysis_reset (RgAnalysisCtx * ctx);
-void rg_analysis_destroy (RgAnalysisCtx * ctx);
-
-G_END_DECLS
-
-#endif /* __RG_ANALYSIS_H__ */
generated by cgit v1.2.1 (git 2.18.0) at 2024-07-01 05:52:30 +0000