+2008-07-19 Jan Schmidt <jan.schmidt@sun.com>
+
+ * 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
+
2008-07-18 Sebastian Dröge <sebastian.droege@collabora.co.uk>
* configure.ac:
$(top_srcdir)/gst/deinterlace/gstdeinterlace.h \
$(top_srcdir)/gst/dvdspu/gstdvdspu.h \
$(top_srcdir)/gst/festival/gstfestival.h \
- $(top_srcdir)/gst/interleave/interleave.h \
- $(top_srcdir)/gst/interleave/deinterleave.h \
$(top_srcdir)/gst/modplug/gstmodplug.h \
$(top_srcdir)/gst/nuvdemux/gstnuvdemux.h \
$(top_srcdir)/gst/rawparse/gstaudioparse.h \
$(top_srcdir)/gst/rawparse/gstvideoparse.h \
- $(top_srcdir)/gst/replaygain/gstrganalysis.h \
- $(top_srcdir)/gst/replaygain/gstrglimiter.h \
- $(top_srcdir)/gst/replaygain/gstrgvolume.h \
$(top_srcdir)/gst/rtpmanager/gstrtpbin.h \
$(top_srcdir)/gst/rtpmanager/gstrtpclient.h \
$(top_srcdir)/gst/rtpmanager/gstrtpjitterbuffer.h \
<xi:include href="xml/element-amrwbparse.xml" />
<xi:include href="xml/element-audioparse.xml" />
<xi:include href="xml/element-deinterlace.xml" />
- <xi:include href="xml/element-deinterleave.xml" />
<xi:include href="xml/element-dfbvideosink.xml" />
<xi:include href="xml/element-dvbsrc.xml" />
<xi:include href="xml/element-dvdspu.xml" />
<xi:include href="xml/element-gstrtpsession.xml" />
<xi:include href="xml/element-gstrtpssrcdemux.xml" />
<xi:include href="xml/element-input-selector.xml" />
- <xi:include href="xml/element-interleave.xml" />
<xi:include href="xml/element-ivorbisdec.xml" />
<xi:include href="xml/element-jackaudiosink.xml" />
<xi:include href="xml/element-metadatademux.xml" />
<xi:include href="xml/element-mythtvsrc.xml" />
<xi:include href="xml/element-nuvdemux.xml" />
<xi:include href="xml/element-output-selector.xml" />
- <xi:include href="xml/element-rganalysis.xml" />
- <xi:include href="xml/element-rglimiter.xml" />
- <xi:include href="xml/element-rgvolume.xml" />
<xi:include href="xml/element-sdlaudiosink.xml" />
<xi:include href="xml/element-sdlvideosink.xml" />
<xi:include href="xml/element-sdpdemux.xml" />
<xi:include href="xml/plugin-gstinterlace.xml" />
<xi:include href="xml/plugin-gstrtpmanager.xml" />
<xi:include href="xml/plugin-h264parse.xml" />
- <xi:include href="xml/plugin-interleave.xml" />
<xi:include href="xml/plugin-jack.xml" />
<xi:include href="xml/plugin-ladspa.xml" />
<xi:include href="xml/plugin-metadata.xml" />
<xi:include href="xml/plugin-nuvdemux.xml" />
<xi:include href="xml/plugin-rawparse.xml" />
<xi:include href="xml/plugin-real.xml" />
- <xi:include href="xml/plugin-replaygain.xml" />
<xi:include href="xml/plugin-rfbsrc.xml" />
<xi:include href="xml/plugin-sdl.xml" />
<xi:include href="xml/plugin-sdp.xml" />
FESTIVAL_DEFAULT_TEXT_MODE
</SECTION>
+<SECTION>
<FILE>element-input-selector</FILE>
<TITLE>input-selector</TITLE>
GstInputSelector
gst_ivorbis_dec_get_type
</SECTION>
-<SECTION>
-<FILE>element-interleave</FILE>
-<TITLE>interleave</TITLE>
-GstInterleave
-<SUBSECTION Standard>
-GST_INTERLEAVE
-GST_INTERLEAVE_CLASS
-GST_INTERLEAVE_GET_CLASS
-GST_IS_INTERLEAVE
-GST_IS_INTERLEAVE_CLASS
-GST_TYPE_INTERLEAVE
-GstInterleaveClass
-GstInterleaveFunc
-gst_interleave_get_type
-</SECTION>
-
-<SECTION>
-<FILE>element-deinterleave</FILE>
-<TITLE>deinterleave</TITLE>
-GstDeinterleave
-<SUBSECTION Standard>
-GST_DEINTERLEAVE
-GST_DEINTERLEAVE_CLASS
-GST_DEINTERLEAVE_GET_CLASS
-GST_IS_DEINTERLEAVE
-GST_IS_DEINTERLEAVE_CLASS
-GST_TYPE_DEINTERLEAVE
-GstDeinterleaveClass
-GstDeinterleaveFunc
-gst_deinterleave_get_type
-</SECTION>
-
<SECTION>
<FILE>element-jackaudiosink</FILE>
GstJackAudioSink
gst_output_selector_get_type
</SECTION>
-<SECTION>
-<FILE>element-rganalysis</FILE>
-<TITLE>rganalysis</TITLE>
-GstRgAnalysis
-<SUBSECTION Standard>
-GstRgAnalysisClass
-GST_RG_ANALYSIS
-GST_RG_ANALYSIS_CLASS
-GST_IS_RG_ANALYSIS
-GST_IS_RG_ANALYSIS_CLASS
-GST_TYPE_RG_ANALYSIS
-gst_rg_analysis_get_type
-</SECTION>
-
-<SECTION>
-<FILE>element-rglimiter</FILE>
-<TITLE>rglimiter</TITLE>
-GstRgLimiter
-<SUBSECTION Standard>
-GstRgLimiterClass
-GST_RG_LIMITER
-GST_RG_LIMITER_CLASS
-GST_IS_RG_LIMITER
-GST_IS_RG_LIMITER_CLASS
-GST_TYPE_RG_LIMITER
-gst_rg_limiter_get_type
-</SECTION>
-
-<SECTION>
-<FILE>element-rgvolume</FILE>
-<TITLE>rgvolume</TITLE>
-GstRgVolume
-<SUBSECTION Standard>
-GstRgVolumeClass
-GST_RG_VOLUME
-GST_RG_VOLUME_CLASS
-GST_IS_RG_VOLUME
-GST_TYPE_RG_VOLUME
-GST_IS_RG_VOLUME_CLASS
-gst_rg_volume_get_type
-</SECTION>
-
<SECTION>
<FILE>element-gstrtpbin</FILE>
<TITLE>gstrtpbin</TITLE>
GstTagSetter GstObject GstElement
GstImplementsInterface GstObject GstElement
GstXOverlay GstObject GstImplementsInterface GstElement
+GstTagSetter GstObject GstElement
+GstColorBalance GstObject GstImplementsInterface GstElement
GstMixer GstObject GstImplementsInterface GstElement
+++ /dev/null
-<plugin>
- <name>interleave</name>
- <description>Audio interleaver/deinterleaver</description>
- <filename>../../gst/interleave/.libs/libgstinterleave.so</filename>
- <basename>libgstinterleave.so</basename>
- <version>0.10.7.1</version>
- <license>LGPL</license>
- <source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins CVS/prerelease</package>
- <origin>http://gstreamer.freedesktop.org</origin>
- <elements>
- <element>
- <name>deinterleave</name>
- <longname>Audio deinterleaver</longname>
- <class>Filter/Converter/Audio</class>
- <description>Splits one interleaved multichannel audio stream into many mono audio streams</description>
- <author>Andy Wingo <wingo at pobox.com>, Iain <iain@prettypeople.org>, Sebastian Dröge <slomo@circular-chaos.org></author>
- <pads>
- <caps>
- <name>sink</name>
- <direction>sink</direction>
- <presence>always</presence>
- <details>audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int){ 1234, 4321 }, width=(int){ 8, 16, 24, 32 }, depth=(int)[ 1, 32 ], signed=(boolean){ true, false }; audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int){ 1234, 4321 }, width=(int){ 32, 64 }</details>
- </caps>
- <caps>
- <name>src%d</name>
- <direction>source</direction>
- <presence>sometimes</presence>
- <details>audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)1, endianness=(int){ 1234, 4321 }, width=(int){ 8, 16, 24, 32 }, depth=(int)[ 1, 32 ], signed=(boolean){ true, false }; audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)1, endianness=(int){ 1234, 4321 }, width=(int){ 32, 64 }</details>
- </caps>
- </pads>
- </element>
- <element>
- <name>interleave</name>
- <longname>Audio interleaver</longname>
- <class>Filter/Converter/Audio</class>
- <description>Folds many mono channels into one interleaved audio stream</description>
- <author>Andy Wingo <wingo at pobox.com>, Sebastian Dröge <slomo@circular-chaos.org></author>
- <pads>
- <caps>
- <name>sink%d</name>
- <direction>sink</direction>
- <presence>request</presence>
- <details>audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)1, endianness=(int){ 1234, 4321 }, width=(int){ 8, 16, 24, 32 }, depth=(int)[ 1, 32 ], signed=(boolean)true; audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)1, endianness=(int){ 1234, 4321 }, width=(int){ 32, 64 }</details>
- </caps>
- <caps>
- <name>src</name>
- <direction>source</direction>
- <presence>always</presence>
- <details>audio/x-raw-int, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int){ 1234, 4321 }, width=(int){ 8, 16, 24, 32 }, depth=(int)[ 1, 32 ], signed=(boolean)true; audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int){ 1234, 4321 }, width=(int){ 32, 64 }</details>
- </caps>
- </pads>
- </element>
- </elements>
-</plugin>
\ No newline at end of file
+++ /dev/null
-<plugin>
- <name>replaygain</name>
- <description>ReplayGain volume normalization</description>
- <filename>../../gst/replaygain/.libs/libgstreplaygain.so</filename>
- <basename>libgstreplaygain.so</basename>
- <version>0.10.7.1</version>
- <license>LGPL</license>
- <source>gst-plugins-bad</source>
- <package>GStreamer Bad Plug-ins CVS/prerelease</package>
- <origin>http://gstreamer.freedesktop.org</origin>
- <elements>
- <element>
- <name>rganalysis</name>
- <longname>ReplayGain analysis</longname>
- <class>Filter/Analyzer/Audio</class>
- <description>Perform the ReplayGain analysis</description>
- <author>René Stadler <mail@renestadler.de></author>
- <pads>
- <caps>
- <name>src</name>
- <direction>source</direction>
- <presence>always</presence>
- <details>audio/x-raw-float, width=(int)32, endianness=(int)1234, channels=(int){ 1, 2 }, rate=(int){ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }; audio/x-raw-int, width=(int)16, depth=(int)[ 1, 16 ], signed=(boolean)true, endianness=(int)1234, channels=(int){ 1, 2 }, rate=(int){ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }</details>
- </caps>
- <caps>
- <name>sink</name>
- <direction>sink</direction>
- <presence>always</presence>
- <details>audio/x-raw-float, width=(int)32, endianness=(int)1234, channels=(int){ 1, 2 }, rate=(int){ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }; audio/x-raw-int, width=(int)16, depth=(int)[ 1, 16 ], signed=(boolean)true, endianness=(int)1234, channels=(int){ 1, 2 }, rate=(int){ 8000, 11025, 12000, 16000, 22050, 24000, 32000, 44100, 48000 }</details>
- </caps>
- </pads>
- </element>
- <element>
- <name>rglimiter</name>
- <longname>ReplayGain limiter</longname>
- <class>Filter/Effect/Audio</class>
- <description>Apply signal compression to raw audio data</description>
- <author>René Stadler <mail@renestadler.de></author>
- <pads>
- <caps>
- <name>src</name>
- <direction>source</direction>
- <presence>always</presence>
- <details>audio/x-raw-float, width=(int)32, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234</details>
- </caps>
- <caps>
- <name>sink</name>
- <direction>sink</direction>
- <presence>always</presence>
- <details>audio/x-raw-float, width=(int)32, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234</details>
- </caps>
- </pads>
- </element>
- <element>
- <name>rgvolume</name>
- <longname>ReplayGain volume</longname>
- <class>Filter/Effect/Audio</class>
- <description>Apply ReplayGain volume adjustment</description>
- <author>René Stadler <mail@renestadler.de></author>
- <pads>
- <caps>
- <name>src</name>
- <direction>source</direction>
- <presence>always</presence>
- <details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)32; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true</details>
- </caps>
- <caps>
- <name>sink</name>
- <direction>sink</direction>
- <presence>always</presence>
- <details>audio/x-raw-float, rate=(int)[ 1, 2147483647 ], channels=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)32; audio/x-raw-int, channels=(int)[ 1, 2147483647 ], rate=(int)[ 1, 2147483647 ], endianness=(int)1234, width=(int)16, depth=(int)16, signed=(boolean)true</details>
- </caps>
- </pads>
- </element>
- </elements>
-</plugin>
\ No newline at end of file
+++ /dev/null
-
-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
+++ /dev/null
-/* 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;
-}
+++ /dev/null
-/* 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__ */
+++ /dev/null
-/* 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;
- }
-}
+++ /dev/null
-/* 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__ */
+++ /dev/null
-/* 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);
+++ /dev/null
-/* 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__ */
+++ /dev/null
-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
+++ /dev/null
-/* 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;
-}
+++ /dev/null
-/* 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__ */
+++ /dev/null
-/* 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;
-}
+++ /dev/null
-/* 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__ */
+++ /dev/null
-/* 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;
- }
-}
+++ /dev/null
-/* 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__ */
+++ /dev/null
-/* 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);
+++ /dev/null
-/* 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__ */
+++ /dev/null
-/* 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);
-}
+++ /dev/null
-/* 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__ */
$(check_neon) \
$(check_ofa) \
$(check_timidity) \
- elements/deinterleave \
- elements/interleave \
- elements/rganalysis \
- elements/rglimiter \
- elements/rgvolume \
elements/selector \
elements/y4menc
AM_CFLAGS = $(GST_OBJ_CFLAGS) $(GST_CHECK_CFLAGS) $(CHECK_CFLAGS) $(GST_OPTION_CFLAGS)
LDADD = $(GST_OBJ_LIBS) $(GST_CHECK_LIBS) $(CHECK_LIBS)
-elements_deinterleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
-elements_deinterleave_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(LDADD)
-elements_interleave_CFLAGS = $(GST_PLUGINS_BASE_CFLAGS) $(CFLAGS) $(AM_CFLAGS)
-elements_interleave_LDADD = $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) $(LDADD)
-
elements_timidity_CFLAGS = $(GST_BASE_CFLAGS) $(AM_CFLAGS)
elements_timidity_LDADD = $(GST_BASE_LIBS) $(LDADD)
+++ /dev/null
-/* GStreamer unit tests for the interleave element
- * Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * 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 <gst/check/gstcheck.h>
-#include <gst/audio/multichannel.h>
-
-GST_START_TEST (test_create_and_unref)
-{
- GstElement *deinterleave;
-
- deinterleave = gst_element_factory_make ("deinterleave", NULL);
- fail_unless (deinterleave != NULL);
-
- gst_element_set_state (deinterleave, GST_STATE_NULL);
- gst_object_unref (deinterleave);
-}
-
-GST_END_TEST;
-
-static GstPad *mysrcpad, **mysinkpads;
-static gint nsinkpads;
-static GstBus *bus;
-static GstElement *deinterleave;
-
-static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-float, "
- "width = (int) 32, "
- "channels = (int) 1, "
- "rate = (int) {32000, 48000}, " "endianness = (int) BYTE_ORDER"));
-
-static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-float, "
- "width = (int) 32, "
- "channels = (int) { 2, 3 }, "
- "rate = (int) {32000, 48000}, " "endianness = (int) BYTE_ORDER"));
-
-#define CAPS_32khz \
- "audio/x-raw-float, " \
- "width = (int) 32, " \
- "channels = (int) 2, " \
- "rate = (int) 32000, " \
- "endianness = (int) BYTE_ORDER"
-
-#define CAPS_48khz \
- "audio/x-raw-float, " \
- "width = (int) 32, " \
- "channels = (int) 2, " \
- "rate = (int) 48000, " \
- "endianness = (int) BYTE_ORDER"
-
-#define CAPS_48khz_3CH \
- "audio/x-raw-float, " \
- "width = (int) 32, " \
- "channels = (int) 3, " \
- "rate = (int) 48000, " \
- "endianness = (int) BYTE_ORDER"
-
-static GstFlowReturn
-deinterleave_chain_func (GstPad * pad, GstBuffer * buffer)
-{
- gint i;
- gfloat *indata;
-
- fail_unless (GST_IS_BUFFER (buffer));
- fail_unless_equals_int (GST_BUFFER_SIZE (buffer), 48000 * sizeof (gfloat));
- fail_unless (GST_BUFFER_DATA (buffer) != NULL);
-
- indata = (gfloat *) GST_BUFFER_DATA (buffer);
-
- if (strcmp (GST_PAD_NAME (pad), "sink0") == 0) {
- for (i = 0; i < 48000; i++)
- fail_unless_equals_float (indata[i], -1.0);
- } else if (strcmp (GST_PAD_NAME (pad), "sink1") == 0) {
- for (i = 0; i < 48000; i++)
- fail_unless_equals_float (indata[i], 1.0);
- } else {
- g_assert_not_reached ();
- }
-
- gst_buffer_unref (buffer);
-
- return GST_FLOW_OK;
-}
-
-static void
-deinterleave_pad_added (GstElement * src, GstPad * pad, gpointer data)
-{
- gchar *name;
- gint link = GPOINTER_TO_INT (data);
-
- if (nsinkpads >= link)
- return;
-
- name = g_strdup_printf ("sink%d", nsinkpads);
-
- mysinkpads[nsinkpads] =
- gst_pad_new_from_static_template (&sinktemplate, name);
- g_free (name);
- fail_if (mysinkpads[nsinkpads] == NULL);
-
- gst_pad_set_chain_function (mysinkpads[nsinkpads], deinterleave_chain_func);
- fail_unless (gst_pad_link (pad, mysinkpads[nsinkpads]) == GST_PAD_LINK_OK);
- gst_pad_set_active (mysinkpads[nsinkpads], TRUE);
- nsinkpads++;
-}
-
-GST_START_TEST (test_2_channels)
-{
- GstPad *sinkpad;
- gint i;
- GstBuffer *inbuf;
- GstCaps *caps;
- gfloat *indata;
-
- mysinkpads = g_new0 (GstPad *, 2);
- nsinkpads = 0;
-
- deinterleave = gst_element_factory_make ("deinterleave", NULL);
- fail_unless (deinterleave != NULL);
-
- mysrcpad = gst_pad_new_from_static_template (&srctemplate, "src");
- fail_unless (mysrcpad != NULL);
-
- caps = gst_caps_from_string (CAPS_48khz);
- fail_unless (gst_pad_set_caps (mysrcpad, caps));
- gst_pad_use_fixed_caps (mysrcpad);
-
- sinkpad = gst_element_get_static_pad (deinterleave, "sink");
- fail_unless (sinkpad != NULL);
- fail_unless (gst_pad_link (mysrcpad, sinkpad) == GST_PAD_LINK_OK);
- g_object_unref (sinkpad);
-
- g_signal_connect (deinterleave, "pad-added",
- G_CALLBACK (deinterleave_pad_added), GINT_TO_POINTER (2));
-
- bus = gst_bus_new ();
- gst_element_set_bus (deinterleave, bus);
-
- fail_unless (gst_element_set_state (deinterleave,
- GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-
- inbuf = gst_buffer_new_and_alloc (2 * 48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 2 * 48000; i += 2) {
- indata[i] = -1.0;
- indata[i + 1] = 1.0;
- }
- gst_buffer_set_caps (inbuf, caps);
-
- fail_unless (gst_pad_push (mysrcpad, inbuf) == GST_FLOW_OK);
-
- fail_unless (gst_element_set_state (deinterleave,
- GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
-
- for (i = 0; i < nsinkpads; i++)
- g_object_unref (mysinkpads[i]);
- g_free (mysinkpads);
- mysinkpads = NULL;
-
- g_object_unref (deinterleave);
- g_object_unref (bus);
- gst_caps_unref (caps);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_2_channels_1_linked)
-{
- GstPad *sinkpad;
- gint i;
- GstBuffer *inbuf;
- GstCaps *caps;
- gfloat *indata;
-
- nsinkpads = 0;
- mysinkpads = g_new0 (GstPad *, 2);
-
- deinterleave = gst_element_factory_make ("deinterleave", NULL);
- fail_unless (deinterleave != NULL);
-
- mysrcpad = gst_pad_new_from_static_template (&srctemplate, "src");
- fail_unless (mysrcpad != NULL);
-
- caps = gst_caps_from_string (CAPS_48khz);
- fail_unless (gst_pad_set_caps (mysrcpad, caps));
- gst_pad_use_fixed_caps (mysrcpad);
-
- sinkpad = gst_element_get_static_pad (deinterleave, "sink");
- fail_unless (sinkpad != NULL);
- fail_unless (gst_pad_link (mysrcpad, sinkpad) == GST_PAD_LINK_OK);
- g_object_unref (sinkpad);
-
- g_signal_connect (deinterleave, "pad-added",
- G_CALLBACK (deinterleave_pad_added), GINT_TO_POINTER (1));
-
- bus = gst_bus_new ();
- gst_element_set_bus (deinterleave, bus);
-
- fail_unless (gst_element_set_state (deinterleave,
- GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-
- inbuf = gst_buffer_new_and_alloc (2 * 48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 2 * 48000; i += 2) {
- indata[i] = -1.0;
- indata[i + 1] = 1.0;
- }
- gst_buffer_set_caps (inbuf, caps);
-
- fail_unless (gst_pad_push (mysrcpad, inbuf) == GST_FLOW_OK);
-
- fail_unless (gst_element_set_state (deinterleave,
- GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
-
- for (i = 0; i < nsinkpads; i++)
- g_object_unref (mysinkpads[i]);
- g_free (mysinkpads);
- mysinkpads = NULL;
-
- g_object_unref (deinterleave);
- g_object_unref (bus);
- gst_caps_unref (caps);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_2_channels_caps_change)
-{
- GstPad *sinkpad;
- GstCaps *caps, *caps2;
- gint i;
- GstBuffer *inbuf;
- gfloat *indata;
-
- nsinkpads = 0;
- mysinkpads = g_new0 (GstPad *, 2);
-
- deinterleave = gst_element_factory_make ("deinterleave", NULL);
- fail_unless (deinterleave != NULL);
-
- mysrcpad = gst_pad_new_from_static_template (&srctemplate, "src");
- fail_unless (mysrcpad != NULL);
-
- caps = gst_caps_from_string (CAPS_48khz);
- fail_unless (gst_pad_set_caps (mysrcpad, caps));
- gst_pad_use_fixed_caps (mysrcpad);
-
- sinkpad = gst_element_get_static_pad (deinterleave, "sink");
- fail_unless (sinkpad != NULL);
- fail_unless (gst_pad_link (mysrcpad, sinkpad) == GST_PAD_LINK_OK);
- g_object_unref (sinkpad);
-
- g_signal_connect (deinterleave, "pad-added",
- G_CALLBACK (deinterleave_pad_added), GINT_TO_POINTER (2));
-
- bus = gst_bus_new ();
- gst_element_set_bus (deinterleave, bus);
-
- fail_unless (gst_element_set_state (deinterleave,
- GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-
- inbuf = gst_buffer_new_and_alloc (2 * 48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 2 * 48000; i += 2) {
- indata[i] = -1.0;
- indata[i + 1] = 1.0;
- }
- gst_buffer_set_caps (inbuf, caps);
-
- fail_unless (gst_pad_push (mysrcpad, inbuf) == GST_FLOW_OK);
-
- caps2 = gst_caps_from_string (CAPS_32khz);
- gst_pad_set_caps (mysrcpad, caps2);
-
- inbuf = gst_buffer_new_and_alloc (2 * 48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 2 * 48000; i += 2) {
- indata[i] = -1.0;
- indata[i + 1] = 1.0;
- }
- gst_buffer_set_caps (inbuf, caps2);
-
- /* Should work fine because the caps changed in a compatible way */
- fail_unless (gst_pad_push (mysrcpad, inbuf) == GST_FLOW_OK);
-
- gst_caps_unref (caps2);
-
- caps2 = gst_caps_from_string (CAPS_48khz_3CH);
- gst_pad_set_caps (mysrcpad, caps2);
-
- inbuf = gst_buffer_new_and_alloc (3 * 48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 3 * 48000; i += 3) {
- indata[i] = -1.0;
- indata[i + 1] = 1.0;
- indata[i + 2] = 0.0;
- }
- gst_buffer_set_caps (inbuf, caps2);
-
- /* Should break because the caps changed in an incompatible way */
- fail_if (gst_pad_push (mysrcpad, inbuf) == GST_FLOW_OK);
-
- fail_unless (gst_element_set_state (deinterleave,
- GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS);
-
- for (i = 0; i < nsinkpads; i++)
- g_object_unref (mysinkpads[i]);
- g_free (mysinkpads);
- mysinkpads = NULL;
-
- g_object_unref (deinterleave);
- g_object_unref (bus);
- gst_caps_unref (caps);
- gst_caps_unref (caps2);
-}
-
-GST_END_TEST;
-
-
-#define SAMPLES_PER_BUFFER 10
-#define NUM_CHANNELS 8
-#define SAMPLE_RATE 44100
-
-static guint pads_created;
-
-static void
-set_channel_positions (GstCaps * caps, int channels,
- GstAudioChannelPosition * channelpositions)
-{
- GValue chanpos = { 0 };
- GValue pos = { 0 };
- GstStructure *structure = gst_caps_get_structure (caps, 0);
- int c;
-
- g_value_init (&chanpos, GST_TYPE_ARRAY);
- g_value_init (&pos, GST_TYPE_AUDIO_CHANNEL_POSITION);
-
- for (c = 0; c < channels; c++) {
- g_value_set_enum (&pos, channelpositions[c]);
- gst_value_array_append_value (&chanpos, &pos);
- }
- g_value_unset (&pos);
-
- gst_structure_set_value (structure, "channel-positions", &chanpos);
- g_value_unset (&chanpos);
-}
-
-static void
-src_handoff_float32_8ch (GstElement * src, GstBuffer * buf, GstPad * pad,
- gpointer user_data)
-{
- GstAudioChannelPosition layout[NUM_CHANNELS];
- GstCaps *caps;
- gfloat *data;
- guint size, i, c;
-
- caps = gst_caps_new_simple ("audio/x-raw-float",
- "width", G_TYPE_INT, 32,
- "depth", G_TYPE_INT, 32,
- "channels", G_TYPE_INT, NUM_CHANNELS,
- "rate", G_TYPE_INT, SAMPLE_RATE,
- "endianness", G_TYPE_INT, G_BYTE_ORDER, NULL);
-
- for (i = 0; i < NUM_CHANNELS; ++i)
- layout[i] = GST_AUDIO_CHANNEL_POSITION_NONE;
-
- set_channel_positions (caps, NUM_CHANNELS, layout);
-
- size = sizeof (gfloat) * SAMPLES_PER_BUFFER * NUM_CHANNELS;
- data = (gfloat *) g_malloc (size);
-
- GST_BUFFER_MALLOCDATA (buf) = (guint8 *) data;
- GST_BUFFER_DATA (buf) = (guint8 *) data;
- GST_BUFFER_SIZE (buf) = size;
-
- GST_BUFFER_OFFSET (buf) = 0;
- GST_BUFFER_TIMESTAMP (buf) = 0;
-
- for (i = 0; i < SAMPLES_PER_BUFFER; ++i) {
- for (c = 0; c < NUM_CHANNELS; ++c) {
- *data = (gfloat) ((i * NUM_CHANNELS) + c);
- ++data;
- }
- }
-
- gst_buffer_set_caps (buf, caps);
- gst_caps_unref (caps);
-}
-
-static gboolean
-float_buffer_check_probe (GstPad * pad, GstBuffer * buf, gpointer userdata)
-{
- gfloat *data;
- guint padnum, numpads;
- guint num, i;
- GstCaps *caps;
- GstStructure *s;
- GstAudioChannelPosition *pos;
- gint channels;
-
- fail_unless_equals_int (sscanf (GST_PAD_NAME (pad), "src%u", &padnum), 1);
-
- numpads = pads_created;
-
- /* Check caps */
- caps = GST_BUFFER_CAPS (buf);
- fail_unless (caps != NULL);
- s = gst_caps_get_structure (caps, 0);
- fail_unless (gst_structure_get_int (s, "channels", &channels));
- fail_unless_equals_int (channels, 1);
- fail_unless (gst_structure_has_field (s, "channel-positions"));
- pos = gst_audio_get_channel_positions (s);
- fail_unless (pos != NULL && pos[0] == GST_AUDIO_CHANNEL_POSITION_NONE);
- g_free (pos);
-
- data = (gfloat *) GST_BUFFER_DATA (buf);
- num = GST_BUFFER_SIZE (buf) / sizeof (gfloat);
-
- /* Check buffer content */
- for (i = 0; i < num; ++i) {
- guint val, rest;
-
- val = (guint) data[i];
- GST_LOG ("%s[%u]: %8f", GST_PAD_NAME (pad), i, data[i]);
- /* can't use the modulo operator in the assertion statement, since due to
- * the way it gets expanded it would be interpreted as a printf operator
- * in the failure case, which will result in segfaults */
- rest = val % numpads;
- /* check that the first channel is on pad src0, the second on src1 etc. */
- fail_unless_equals_int (rest, padnum);
- }
-
- return TRUE; /* don't drop data */
-}
-
-static void
-pad_added_setup_data_check_float32_8ch_cb (GstElement * deinterleave,
- GstPad * pad, GstElement * pipeline)
-{
- GstElement *queue, *sink;
- GstPad *sinkpad;
-
- queue = gst_element_factory_make ("queue", NULL);
- fail_unless (queue != NULL);
-
- sink = gst_element_factory_make ("fakesink", NULL);
- fail_unless (sink != NULL);
-
- gst_bin_add_many (GST_BIN (pipeline), queue, sink, NULL);
- fail_unless (gst_element_link_many (queue, sink, NULL));
-
- sinkpad = gst_element_get_static_pad (queue, "sink");
- fail_unless_equals_int (gst_pad_link (pad, sinkpad), GST_PAD_LINK_OK);
- gst_object_unref (sinkpad);
-
- gst_pad_add_buffer_probe (pad, G_CALLBACK (float_buffer_check_probe), NULL);
-
- gst_element_set_state (sink, GST_STATE_PLAYING);
- gst_element_set_state (queue, GST_STATE_PLAYING);
-
- GST_LOG ("new pad: %s", GST_PAD_NAME (pad));
- ++pads_created;
-}
-
-static GstElement *
-make_fake_src_8chans_float32 (void)
-{
- GstElement *src;
-
- src = gst_element_factory_make ("fakesrc", "src");
- fail_unless (src != NULL, "failed to create fakesrc element");
-
- g_object_set (src, "num-buffers", 1, NULL);
- g_object_set (src, "signal-handoffs", TRUE, NULL);
-
- g_signal_connect (src, "handoff", G_CALLBACK (src_handoff_float32_8ch), NULL);
-
- return src;
-}
-
-GST_START_TEST (test_8_channels_float32)
-{
- GstElement *pipeline, *src, *deinterleave;
- GstMessage *msg;
-
- pipeline = (GstElement *) gst_pipeline_new ("pipeline");
- fail_unless (pipeline != NULL, "failed to create pipeline");
-
- src = make_fake_src_8chans_float32 ();
-
- deinterleave = gst_element_factory_make ("deinterleave", "deinterleave");
- fail_unless (deinterleave != NULL, "failed to create deinterleave element");
- g_object_set (deinterleave, "keep-positions", TRUE, NULL);
-
- gst_bin_add_many (GST_BIN (pipeline), src, deinterleave, NULL);
-
- fail_unless (gst_element_link (src, deinterleave),
- "failed to link src <=> deinterleave");
-
- g_signal_connect (deinterleave, "pad-added",
- G_CALLBACK (pad_added_setup_data_check_float32_8ch_cb), pipeline);
-
- pads_created = 0;
-
- gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
- msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
- gst_message_unref (msg);
-
- fail_unless_equals_int (pads_created, NUM_CHANNELS);
-
- gst_element_set_state (pipeline, GST_STATE_NULL);
- gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-static Suite *
-deinterleave_suite (void)
-{
- Suite *s = suite_create ("deinterleave");
- TCase *tc_chain = tcase_create ("general");
-
- suite_add_tcase (s, tc_chain);
- tcase_add_test (tc_chain, test_create_and_unref);
- tcase_add_test (tc_chain, test_2_channels);
- tcase_add_test (tc_chain, test_2_channels_1_linked);
- tcase_add_test (tc_chain, test_2_channels_caps_change);
- tcase_add_test (tc_chain, test_8_channels_float32);
-
- return s;
-}
-
-GST_CHECK_MAIN (deinterleave);
+++ /dev/null
-/* GStreamer unit tests for the interleave element
- * Copyright (C) 2007 Tim-Philipp Müller <tim centricular net>
- * Copyright (C) 2008 Sebastian Dröge <slomo@circular-chaos.org>
- *
- * 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 <gst/check/gstcheck.h>
-#include <gst/audio/multichannel.h>
-
-GST_START_TEST (test_create_and_unref)
-{
- GstElement *interleave;
-
- interleave = gst_element_factory_make ("interleave", NULL);
- fail_unless (interleave != NULL);
-
- gst_element_set_state (interleave, GST_STATE_NULL);
- gst_object_unref (interleave);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_request_pads)
-{
- GstElement *interleave;
-
- GstPad *pad1, *pad2;
-
- interleave = gst_element_factory_make ("interleave", NULL);
- fail_unless (interleave != NULL);
-
- pad1 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (pad1 != NULL);
- fail_unless_equals_string (GST_OBJECT_NAME (pad1), "sink0");
-
- pad2 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (pad2 != NULL);
- fail_unless_equals_string (GST_OBJECT_NAME (pad2), "sink1");
-
- gst_element_release_request_pad (interleave, pad2);
- gst_object_unref (pad2);
- gst_element_release_request_pad (interleave, pad1);
- gst_object_unref (pad1);
-
- gst_element_set_state (interleave, GST_STATE_NULL);
- gst_object_unref (interleave);
-}
-
-GST_END_TEST;
-
-static GstPad **mysrcpads, *mysinkpad;
-
-static GstBus *bus;
-
-static GstElement *interleave;
-
-static gint have_data;
-
-static gfloat input[2];
-
-static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-float, "
- "width = (int) 32, "
- "channels = (int) 2, "
- "rate = (int) 48000, " "endianness = (int) BYTE_ORDER"));
-
-static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS ("audio/x-raw-float, "
- "width = (int) 32, "
- "channels = (int) 1, "
- "rate = (int) 48000, " "endianness = (int) BYTE_ORDER"));
-
-#define CAPS_48khz \
- "audio/x-raw-float, " \
- "width = (int) 32, " \
- "channels = (int) 1, " \
- "rate = (int) 48000, " \
- "endianness = (int) BYTE_ORDER"
-
-static GstFlowReturn
-interleave_chain_func (GstPad * pad, GstBuffer * buffer)
-{
- gfloat *outdata;
-
- gint i;
-
- fail_unless (GST_IS_BUFFER (buffer));
- fail_unless_equals_int (GST_BUFFER_SIZE (buffer),
- 48000 * 2 * sizeof (gfloat));
- fail_unless (GST_BUFFER_DATA (buffer) != NULL);
-
- outdata = (gfloat *) GST_BUFFER_DATA (buffer);
-
- for (i = 0; i < 48000 * 2; i += 2) {
- fail_unless_equals_float (outdata[i], input[0]);
- fail_unless_equals_float (outdata[i + 1], input[1]);
- }
-
- have_data++;
-
- gst_buffer_unref (buffer);
-
- return GST_FLOW_OK;
-}
-
-GST_START_TEST (test_interleave_2ch)
-{
- GstElement *queue;
-
- GstPad *sink0, *sink1, *src, *tmp;
-
- GstCaps *caps;
-
- gint i;
-
- GstBuffer *inbuf;
-
- gfloat *indata;
-
- mysrcpads = g_new0 (GstPad *, 2);
-
- have_data = 0;
-
- interleave = gst_element_factory_make ("interleave", NULL);
- fail_unless (interleave != NULL);
-
- queue = gst_element_factory_make ("queue", "queue");
- fail_unless (queue != NULL);
-
- sink0 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (sink0 != NULL);
- fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink0");
-
- sink1 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (sink1 != NULL);
- fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink1");
-
- mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0");
- fail_unless (mysrcpads[0] != NULL);
-
- caps = gst_caps_from_string (CAPS_48khz);
- fail_unless (gst_pad_set_caps (mysrcpads[0], caps));
- gst_pad_use_fixed_caps (mysrcpads[0]);
-
- mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1");
- fail_unless (mysrcpads[1] != NULL);
-
- fail_unless (gst_pad_set_caps (mysrcpads[1], caps));
- gst_pad_use_fixed_caps (mysrcpads[1]);
-
- tmp = gst_element_get_static_pad (queue, "sink");
- fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
- tmp = gst_element_get_static_pad (queue, "src");
- fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
-
- fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK);
-
- mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
- fail_unless (mysinkpad != NULL);
- gst_pad_set_chain_function (mysinkpad, interleave_chain_func);
- gst_pad_set_active (mysinkpad, TRUE);
-
- src = gst_element_get_static_pad (interleave, "src");
- fail_unless (src != NULL);
- fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK);
- gst_object_unref (src);
-
- bus = gst_bus_new ();
- gst_element_set_bus (interleave, bus);
-
- fail_unless (gst_element_set_state (interleave,
- GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
- fail_unless (gst_element_set_state (queue,
- GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-
- input[0] = -1.0;
- inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 48000; i++)
- indata[i] = -1.0;
- gst_buffer_set_caps (inbuf, caps);
- fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
-
- input[1] = 1.0;
- inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 48000; i++)
- indata[i] = 1.0;
- gst_buffer_set_caps (inbuf, caps);
- fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
- inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 48000; i++)
- indata[i] = -1.0;
- gst_buffer_set_caps (inbuf, caps);
- fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
-
- inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 48000; i++)
- indata[i] = 1.0;
- gst_buffer_set_caps (inbuf, caps);
- fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
- fail_unless (have_data == 2);
-
- gst_object_unref (mysrcpads[0]);
- gst_object_unref (mysrcpads[1]);
- gst_object_unref (mysinkpad);
-
- gst_element_release_request_pad (interleave, sink0);
- gst_object_unref (sink0);
- gst_element_release_request_pad (interleave, sink1);
- gst_object_unref (sink1);
-
- gst_element_set_state (interleave, GST_STATE_NULL);
- gst_element_set_state (queue, GST_STATE_NULL);
- gst_object_unref (interleave);
- gst_object_unref (queue);
- gst_object_unref (bus);
- gst_caps_unref (caps);
-
- g_free (mysrcpads);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_interleave_2ch_1eos)
-{
- GstElement *queue;
-
- GstPad *sink0, *sink1, *src, *tmp;
-
- GstCaps *caps;
-
- gint i;
-
- GstBuffer *inbuf;
-
- gfloat *indata;
-
- mysrcpads = g_new0 (GstPad *, 2);
-
- have_data = 0;
-
- interleave = gst_element_factory_make ("interleave", NULL);
- fail_unless (interleave != NULL);
-
- queue = gst_element_factory_make ("queue", "queue");
- fail_unless (queue != NULL);
-
- sink0 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (sink0 != NULL);
- fail_unless_equals_string (GST_OBJECT_NAME (sink0), "sink0");
-
- sink1 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (sink1 != NULL);
- fail_unless_equals_string (GST_OBJECT_NAME (sink1), "sink1");
-
- mysrcpads[0] = gst_pad_new_from_static_template (&srctemplate, "src0");
- fail_unless (mysrcpads[0] != NULL);
-
- caps = gst_caps_from_string (CAPS_48khz);
- fail_unless (gst_pad_set_caps (mysrcpads[0], caps));
- gst_pad_use_fixed_caps (mysrcpads[0]);
-
- mysrcpads[1] = gst_pad_new_from_static_template (&srctemplate, "src1");
- fail_unless (mysrcpads[1] != NULL);
-
- fail_unless (gst_pad_set_caps (mysrcpads[1], caps));
- gst_pad_use_fixed_caps (mysrcpads[1]);
-
- tmp = gst_element_get_static_pad (queue, "sink");
- fail_unless (gst_pad_link (mysrcpads[0], tmp) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
- tmp = gst_element_get_static_pad (queue, "src");
- fail_unless (gst_pad_link (tmp, sink0) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
-
- fail_unless (gst_pad_link (mysrcpads[1], sink1) == GST_PAD_LINK_OK);
-
- mysinkpad = gst_pad_new_from_static_template (&sinktemplate, "sink");
- fail_unless (mysinkpad != NULL);
- gst_pad_set_chain_function (mysinkpad, interleave_chain_func);
- gst_pad_set_active (mysinkpad, TRUE);
-
- src = gst_element_get_static_pad (interleave, "src");
- fail_unless (src != NULL);
- fail_unless (gst_pad_link (src, mysinkpad) == GST_PAD_LINK_OK);
- gst_object_unref (src);
-
- bus = gst_bus_new ();
- gst_element_set_bus (interleave, bus);
-
- fail_unless (gst_element_set_state (interleave,
- GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
- fail_unless (gst_element_set_state (queue,
- GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS);
-
- input[0] = -1.0;
- inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 48000; i++)
- indata[i] = -1.0;
- gst_buffer_set_caps (inbuf, caps);
- fail_unless (gst_pad_push (mysrcpads[0], inbuf) == GST_FLOW_OK);
-
- input[1] = 1.0;
- inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 48000; i++)
- indata[i] = 1.0;
- gst_buffer_set_caps (inbuf, caps);
- fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
- input[0] = 0.0;
- gst_pad_push_event (mysrcpads[0], gst_event_new_eos ());
-
- input[1] = 1.0;
- inbuf = gst_buffer_new_and_alloc (48000 * sizeof (gfloat));
- indata = (gfloat *) GST_BUFFER_DATA (inbuf);
- for (i = 0; i < 48000; i++)
- indata[i] = 1.0;
- gst_buffer_set_caps (inbuf, caps);
- fail_unless (gst_pad_push (mysrcpads[1], inbuf) == GST_FLOW_OK);
-
- fail_unless (have_data == 2);
-
- gst_object_unref (mysrcpads[0]);
- gst_object_unref (mysrcpads[1]);
- gst_object_unref (mysinkpad);
-
- gst_element_release_request_pad (interleave, sink0);
- gst_object_unref (sink0);
- gst_element_release_request_pad (interleave, sink1);
- gst_object_unref (sink1);
-
- gst_element_set_state (interleave, GST_STATE_NULL);
- gst_element_set_state (queue, GST_STATE_NULL);
- gst_object_unref (interleave);
- gst_object_unref (queue);
- gst_object_unref (bus);
- gst_caps_unref (caps);
-
- g_free (mysrcpads);
-}
-
-GST_END_TEST;
-
-static void
-src_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
- gpointer user_data)
-{
- gint n = GPOINTER_TO_INT (user_data);
-
- GstCaps *caps;
-
- gfloat *data;
-
- gint i;
-
- if (GST_PAD_CAPS (pad))
- caps = gst_caps_ref (GST_PAD_CAPS (pad));
- else {
- caps = gst_caps_new_simple ("audio/x-raw-float",
- "width", G_TYPE_INT, 32,
- "channels", G_TYPE_INT, 1,
- "rate", G_TYPE_INT, 48000, "endianness", G_TYPE_INT, G_BYTE_ORDER,
- NULL);
-
- if (n == 2) {
- GstAudioChannelPosition pos[1] =
- { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT };
- gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
- } else if (n == 3) {
- GstAudioChannelPosition pos[1] =
- { GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT };
- gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
- }
- }
-
- data = g_new (gfloat, 48000);
- GST_BUFFER_MALLOCDATA (buffer) = (guint8 *) data;
- GST_BUFFER_DATA (buffer) = (guint8 *) data;
- GST_BUFFER_SIZE (buffer) = 48000 * sizeof (gfloat);
-
- GST_BUFFER_OFFSET (buffer) = GST_BUFFER_OFFSET_NONE;
- GST_BUFFER_TIMESTAMP (buffer) = GST_CLOCK_TIME_NONE;
- GST_BUFFER_OFFSET_END (buffer) = GST_BUFFER_OFFSET_NONE;
- GST_BUFFER_DURATION (buffer) = GST_SECOND;
-
- gst_buffer_set_caps (buffer, caps);
- gst_caps_unref (caps);
-
- for (i = 0; i < 48000; i++)
- data[i] = (n % 2 == 0) ? -1.0 : 1.0;
-}
-
-static void
-sink_handoff_float32 (GstElement * element, GstBuffer * buffer, GstPad * pad,
- gpointer user_data)
-{
- gint i;
-
- gfloat *data;
-
- GstCaps *caps;
-
- gint n = GPOINTER_TO_INT (user_data);
-
- fail_unless (GST_IS_BUFFER (buffer));
- fail_unless_equals_int (GST_BUFFER_SIZE (buffer),
- 48000 * 2 * sizeof (gfloat));
- fail_unless_equals_int (GST_BUFFER_DURATION (buffer), GST_SECOND);
-
- caps = gst_caps_new_simple ("audio/x-raw-float",
- "width", G_TYPE_INT, 32,
- "channels", G_TYPE_INT, 2,
- "rate", G_TYPE_INT, 48000, "endianness", G_TYPE_INT, G_BYTE_ORDER, NULL);
-
- if (n == 0) {
- GstAudioChannelPosition pos[2] =
- { GST_AUDIO_CHANNEL_POSITION_NONE, GST_AUDIO_CHANNEL_POSITION_NONE };
- gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
- } else if (n == 1) {
- GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_LEFT,
- GST_AUDIO_CHANNEL_POSITION_FRONT_RIGHT
- };
- gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
- } else if (n == 2) {
- GstAudioChannelPosition pos[2] = { GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER,
- GST_AUDIO_CHANNEL_POSITION_REAR_CENTER
- };
- gst_audio_set_channel_positions (gst_caps_get_structure (caps, 0), pos);
- }
-
- fail_unless (gst_caps_is_equal (caps, GST_BUFFER_CAPS (buffer)));
- gst_caps_unref (caps);
-
- data = (gfloat *) GST_BUFFER_DATA (buffer);
-
- for (i = 0; i < 48000 * 2; i += 2) {
- fail_unless_equals_float (data[i], -1.0);
- fail_unless_equals_float (data[i + 1], 1.0);
- }
-
- have_data++;
-}
-
-GST_START_TEST (test_interleave_2ch_pipeline)
-{
- GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
-
- GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
-
- GstMessage *msg;
-
- have_data = 0;
-
- pipeline = (GstElement *) gst_pipeline_new ("pipeline");
- fail_unless (pipeline != NULL);
-
- src1 = gst_element_factory_make ("fakesrc", "src1");
- fail_unless (src1 != NULL);
- g_object_set (src1, "num-buffers", 4, NULL);
- g_object_set (src1, "signal-handoffs", TRUE, NULL);
- g_signal_connect (src1, "handoff", G_CALLBACK (src_handoff_float32),
- GINT_TO_POINTER (0));
- gst_bin_add (GST_BIN (pipeline), src1);
-
- src2 = gst_element_factory_make ("fakesrc", "src2");
- fail_unless (src2 != NULL);
- g_object_set (src2, "num-buffers", 4, NULL);
- g_object_set (src2, "signal-handoffs", TRUE, NULL);
- g_signal_connect (src2, "handoff", G_CALLBACK (src_handoff_float32),
- GINT_TO_POINTER (1));
- gst_bin_add (GST_BIN (pipeline), src2);
-
- queue = gst_element_factory_make ("queue", "queue");
- fail_unless (queue != NULL);
- gst_bin_add (GST_BIN (pipeline), queue);
-
- interleave = gst_element_factory_make ("interleave", "interleave");
- fail_unless (interleave != NULL);
- gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
-
- sinkpad0 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (sinkpad0 != NULL);
- tmp = gst_element_get_static_pad (src1, "src");
- fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
-
- sinkpad1 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (sinkpad1 != NULL);
- tmp = gst_element_get_static_pad (src2, "src");
- tmp2 = gst_element_get_static_pad (queue, "sink");
- fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
- gst_object_unref (tmp2);
- tmp = gst_element_get_static_pad (queue, "src");
- fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
-
- sink = gst_element_factory_make ("fakesink", "sink");
- fail_unless (sink != NULL);
- g_object_set (sink, "signal-handoffs", TRUE, NULL);
- g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
- GINT_TO_POINTER (0));
- gst_bin_add (GST_BIN (pipeline), sink);
- tmp = gst_element_get_static_pad (interleave, "src");
- tmp2 = gst_element_get_static_pad (sink, "sink");
- fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
- gst_object_unref (tmp2);
-
- gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
- msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
- gst_message_unref (msg);
-
- fail_unless (have_data == 4);
-
- gst_element_set_state (pipeline, GST_STATE_NULL);
- gst_element_release_request_pad (interleave, sinkpad0);
- gst_object_unref (sinkpad0);
- gst_element_release_request_pad (interleave, sinkpad1);
- gst_object_unref (sinkpad1);
- gst_object_unref (interleave);
- gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_interleave_2ch_pipeline_input_chanpos)
-{
- GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
-
- GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
-
- GstMessage *msg;
-
- have_data = 0;
-
- pipeline = (GstElement *) gst_pipeline_new ("pipeline");
- fail_unless (pipeline != NULL);
-
- src1 = gst_element_factory_make ("fakesrc", "src1");
- fail_unless (src1 != NULL);
- g_object_set (src1, "num-buffers", 4, NULL);
- g_object_set (src1, "signal-handoffs", TRUE, NULL);
- g_signal_connect (src1, "handoff", G_CALLBACK (src_handoff_float32),
- GINT_TO_POINTER (2));
- gst_bin_add (GST_BIN (pipeline), src1);
-
- src2 = gst_element_factory_make ("fakesrc", "src2");
- fail_unless (src2 != NULL);
- g_object_set (src2, "num-buffers", 4, NULL);
- g_object_set (src2, "signal-handoffs", TRUE, NULL);
- g_signal_connect (src2, "handoff", G_CALLBACK (src_handoff_float32),
- GINT_TO_POINTER (3));
- gst_bin_add (GST_BIN (pipeline), src2);
-
- queue = gst_element_factory_make ("queue", "queue");
- fail_unless (queue != NULL);
- gst_bin_add (GST_BIN (pipeline), queue);
-
- interleave = gst_element_factory_make ("interleave", "interleave");
- fail_unless (interleave != NULL);
- g_object_set (interleave, "channel-positions-from-input", TRUE, NULL);
- gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
-
- sinkpad0 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (sinkpad0 != NULL);
- tmp = gst_element_get_static_pad (src1, "src");
- fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
-
- sinkpad1 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (sinkpad1 != NULL);
- tmp = gst_element_get_static_pad (src2, "src");
- tmp2 = gst_element_get_static_pad (queue, "sink");
- fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
- gst_object_unref (tmp2);
- tmp = gst_element_get_static_pad (queue, "src");
- fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
-
- sink = gst_element_factory_make ("fakesink", "sink");
- fail_unless (sink != NULL);
- g_object_set (sink, "signal-handoffs", TRUE, NULL);
- g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
- GINT_TO_POINTER (1));
- gst_bin_add (GST_BIN (pipeline), sink);
- tmp = gst_element_get_static_pad (interleave, "src");
- tmp2 = gst_element_get_static_pad (sink, "sink");
- fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
- gst_object_unref (tmp2);
-
- gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
- msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
- gst_message_unref (msg);
-
- fail_unless (have_data == 4);
-
- gst_element_set_state (pipeline, GST_STATE_NULL);
- gst_element_release_request_pad (interleave, sinkpad0);
- gst_object_unref (sinkpad0);
- gst_element_release_request_pad (interleave, sinkpad1);
- gst_object_unref (sinkpad1);
- gst_object_unref (interleave);
- gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_interleave_2ch_pipeline_custom_chanpos)
-{
- GstElement *pipeline, *queue, *src1, *src2, *interleave, *sink;
-
- GstPad *sinkpad0, *sinkpad1, *tmp, *tmp2;
-
- GstMessage *msg;
-
- GValueArray *arr;
- GValue val = { 0, };
-
- have_data = 0;
-
- pipeline = (GstElement *) gst_pipeline_new ("pipeline");
- fail_unless (pipeline != NULL);
-
- src1 = gst_element_factory_make ("fakesrc", "src1");
- fail_unless (src1 != NULL);
- g_object_set (src1, "num-buffers", 4, NULL);
- g_object_set (src1, "signal-handoffs", TRUE, NULL);
- g_signal_connect (src1, "handoff", G_CALLBACK (src_handoff_float32),
- GINT_TO_POINTER (0));
- gst_bin_add (GST_BIN (pipeline), src1);
-
- src2 = gst_element_factory_make ("fakesrc", "src2");
- fail_unless (src2 != NULL);
- g_object_set (src2, "num-buffers", 4, NULL);
- g_object_set (src2, "signal-handoffs", TRUE, NULL);
- g_signal_connect (src2, "handoff", G_CALLBACK (src_handoff_float32),
- GINT_TO_POINTER (1));
- gst_bin_add (GST_BIN (pipeline), src2);
-
- queue = gst_element_factory_make ("queue", "queue");
- fail_unless (queue != NULL);
- gst_bin_add (GST_BIN (pipeline), queue);
-
- interleave = gst_element_factory_make ("interleave", "interleave");
- fail_unless (interleave != NULL);
- g_object_set (interleave, "channel-positions-from-input", FALSE, NULL);
- arr = g_value_array_new (2);
- g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION);
- g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_FRONT_CENTER);
- g_value_array_append (arr, &val);
- g_value_reset (&val);
- g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_REAR_CENTER);
- g_value_array_append (arr, &val);
- g_value_unset (&val);
- g_object_set (interleave, "channel-positions", arr, NULL);
- g_value_array_free (arr);
- gst_bin_add (GST_BIN (pipeline), gst_object_ref (interleave));
-
- sinkpad0 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (sinkpad0 != NULL);
- tmp = gst_element_get_static_pad (src1, "src");
- fail_unless (gst_pad_link (tmp, sinkpad0) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
-
- sinkpad1 = gst_element_get_request_pad (interleave, "sink%d");
- fail_unless (sinkpad1 != NULL);
- tmp = gst_element_get_static_pad (src2, "src");
- tmp2 = gst_element_get_static_pad (queue, "sink");
- fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
- gst_object_unref (tmp2);
- tmp = gst_element_get_static_pad (queue, "src");
- fail_unless (gst_pad_link (tmp, sinkpad1) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
-
- sink = gst_element_factory_make ("fakesink", "sink");
- fail_unless (sink != NULL);
- g_object_set (sink, "signal-handoffs", TRUE, NULL);
- g_signal_connect (sink, "handoff", G_CALLBACK (sink_handoff_float32),
- GINT_TO_POINTER (2));
- gst_bin_add (GST_BIN (pipeline), sink);
- tmp = gst_element_get_static_pad (interleave, "src");
- tmp2 = gst_element_get_static_pad (sink, "sink");
- fail_unless (gst_pad_link (tmp, tmp2) == GST_PAD_LINK_OK);
- gst_object_unref (tmp);
- gst_object_unref (tmp2);
-
- gst_element_set_state (pipeline, GST_STATE_PLAYING);
-
- msg = gst_bus_poll (GST_ELEMENT_BUS (pipeline), GST_MESSAGE_EOS, -1);
- gst_message_unref (msg);
-
- fail_unless (have_data == 4);
-
- gst_element_set_state (pipeline, GST_STATE_NULL);
- gst_element_release_request_pad (interleave, sinkpad0);
- gst_object_unref (sinkpad0);
- gst_element_release_request_pad (interleave, sinkpad1);
- gst_object_unref (sinkpad1);
- gst_object_unref (interleave);
- gst_object_unref (pipeline);
-}
-
-GST_END_TEST;
-
-static Suite *
-interleave_suite (void)
-{
- Suite *s = suite_create ("interleave");
-
- TCase *tc_chain = tcase_create ("general");
-
- suite_add_tcase (s, tc_chain);
- tcase_add_test (tc_chain, test_create_and_unref);
- tcase_add_test (tc_chain, test_request_pads);
- tcase_add_test (tc_chain, test_interleave_2ch);
- tcase_add_test (tc_chain, test_interleave_2ch_1eos);
- tcase_add_test (tc_chain, test_interleave_2ch_pipeline);
- tcase_add_test (tc_chain, test_interleave_2ch_pipeline_input_chanpos);
- tcase_add_test (tc_chain, test_interleave_2ch_pipeline_custom_chanpos);
-
- return s;
-}
-
-GST_CHECK_MAIN (interleave);
+++ /dev/null
-/* GStreamer ReplayGain analysis
- *
- * Copyright (C) 2006 Rene Stadler <mail@renestadler.de>
- *
- * rganalysis.c: Unit test for the rganalysis element
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-/* Some things to note about the RMS window length of the analysis algorithm and
- * thus the implementation used in the element: Processing divides input data
- * into 50ms windows at some point. Some details about this that normally do
- * not matter:
- *
- * 1. At the end of a stream, the remainder of data that did not fill up the
- * last 50ms window is simply discarded.
- *
- * 2. If the sample rate changes during a stream, the currently running window
- * is discarded and the equal loudness filter gets reset as if a new stream
- * started.
- *
- * 3. For the album gain, it is not entirely correct to think of obtaining it
- * like "as if all the tracks are analyzed as one track". There isn't a
- * separate window being tracked for album processing, so at stream (track)
- * end, the remaining unfilled window does not contribute to the album gain
- * either.
- *
- * 4. If a waveform with a result gain G is concatenated to itself and the
- * result processed as a track, the gain can be different from G if and only
- * if the duration of the original waveform is not an integer multiple of
- * 50ms. If the original waveform gets processed as a single track and then
- * the same data again as a subsequent track, the album result gain will
- * always match G (this is implied by 3.).
- *
- * 5. A stream shorter than 50ms cannot be analyzed. At 8000 and 48000 Hz,
- * this corresponds to 400 resp. 2400 frames. If a stream is shorter than
- * 50ms, the element will not generate tags at EOS (only if an album
- * finished, but only album tags are generated then). This is not an
- * erroneous condition, the element should behave normally.
- *
- * The limitations outlined in 1.-4. do not apply to the peak values. Every
- * single sample is accounted for when looking for the peak. Thus the album
- * peak is guaranteed to be the maximum value of all track peaks.
- *
- * In normal day-to-day use, these little facts are unlikely to be relevant, but
- * they have to be kept in mind for writing the tests here.
- */
-
-#include <gst/check/gstcheck.h>
-
-GList *buffers = NULL;
-
-/* For ease of programming we use globals to keep refs for our floating src and
- * sink pads we create; otherwise we always have to do get_pad, get_peer, and
- * then remove references in every test function */
-static GstPad *mysrcpad, *mysinkpad;
-
-/* Mapping from supported sample rates to the correct result gain for the
- * following test waveform: 20 * 512 samples with a quarter-full amplitude of
- * toggling sign, changing every 48 samples and starting with the positive
- * value.
- *
- * Even if we would generate a wave describing a signal with the same frequency
- * at each sampling rate, the results would vary (slightly). Hence the simple
- * generation method, since we cannot use a constant value as expected result
- * anyways. For all sample rates, changing the sign every 48 frames gives a
- * sane frequency. Buffers containing data that forms such a waveform is
- * created using the test_buffer_square_{float,int16}_{mono,stereo} functions
- * below.
- *
- * The results have been checked against what the metaflac and wavegain programs
- * generate for such a stream. If you want to verify these, be sure that the
- * metaflac program does not produce incorrect results in your environment: I
- * found a strange bug in the (defacto) reference code for the analysis that
- * sometimes leads to incorrect RMS window lengths. */
-
-struct rate_test
-{
- guint sample_rate;
- gdouble gain;
-};
-
-static const struct rate_test supported_rates[] = {
- {8000, -0.91},
- {11025, -2.80},
- {12000, -3.13},
- {16000, -4.26},
- {22050, -5.64},
- {24000, -5.87},
- {32000, -6.03},
- {44100, -6.20},
- {48000, -6.14}
-};
-
-/* Lookup the correct gain adjustment result in above array. */
-
-static gdouble
-get_expected_gain (guint sample_rate)
-{
- gint i;
-
- for (i = G_N_ELEMENTS (supported_rates); i--;)
- if (supported_rates[i].sample_rate == sample_rate)
- return supported_rates[i].gain;
- g_return_val_if_reached (0.0);
-}
-
-#define SILENCE_GAIN 64.82
-
-#define REPLAY_GAIN_CAPS \
- "channels = (int) { 1, 2 }, " \
- "rate = (int) { 8000, 11025, 12000, 16000, 22050, " \
- "24000, 32000, 44100, 48000 }"
-
-#define RG_ANALYSIS_CAPS_TEMPLATE_STRING \
- "audio/x-raw-float, " \
- "width = (int) 32, " \
- "endianness = (int) BYTE_ORDER, " \
- REPLAY_GAIN_CAPS \
- "; " \
- "audio/x-raw-int, " \
- "width = (int) 16, " \
- "depth = (int) [ 1, 16 ], " \
- "signed = (boolean) true, " \
- "endianness = (int) BYTE_ORDER, " \
- REPLAY_GAIN_CAPS
-
-static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS (RG_ANALYSIS_CAPS_TEMPLATE_STRING)
- );
-static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS (RG_ANALYSIS_CAPS_TEMPLATE_STRING)
- );
-
-GstElement *
-setup_rganalysis ()
-{
- GstElement *analysis;
- GstBus *bus;
-
- GST_DEBUG ("setup_rganalysis");
- analysis = gst_check_setup_element ("rganalysis");
- mysrcpad = gst_check_setup_src_pad (analysis, &srctemplate, NULL);
- mysinkpad = gst_check_setup_sink_pad (analysis, &sinktemplate, NULL);
- gst_pad_set_active (mysrcpad, TRUE);
- gst_pad_set_active (mysinkpad, TRUE);
-
- bus = gst_bus_new ();
- gst_element_set_bus (analysis, bus);
- /* gst_element_set_bus does not steal a reference. */
- gst_object_unref (bus);
-
- return analysis;
-}
-
-void
-cleanup_rganalysis (GstElement * element)
-{
- GST_DEBUG ("cleanup_rganalysis");
-
- g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
- g_list_free (buffers);
- buffers = NULL;
-
- /* The bus owns references to the element: */
- gst_element_set_bus (element, NULL);
-
- gst_pad_set_active (mysrcpad, FALSE);
- gst_pad_set_active (mysinkpad, FALSE);
- gst_check_teardown_src_pad (element);
- gst_check_teardown_sink_pad (element);
- gst_check_teardown_element (element);
-}
-
-static void
-set_playing_state (GstElement * element)
-{
- fail_unless (gst_element_set_state (element,
- GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
- "Could not set state to PLAYING");
-}
-
-static void
-send_eos_event (GstElement * element)
-{
- GstBus *bus = gst_element_get_bus (element);
- GstPad *pad = gst_element_get_static_pad (element, "sink");
- GstEvent *event = gst_event_new_eos ();
-
- fail_unless (gst_pad_send_event (pad, event),
- "Cannot send EOS event: Not handled.");
-
- /* There is no sink element, so _we_ post the EOS message on the bus here. Of
- * course we generate any EOS ourselves, but this allows us to poll for the
- * EOS message in poll_eos if we expect the element to _not_ generate a TAG
- * message. That's better than waiting for a timeout to lapse. */
- fail_unless (gst_bus_post (bus, gst_message_new_eos (NULL)));
-
- gst_object_unref (bus);
- gst_object_unref (pad);
-}
-
-static void
-send_tag_event (GstElement * element, GstTagList * tag_list)
-{
- GstPad *pad = gst_element_get_static_pad (element, "sink");
- GstEvent *event = gst_event_new_tag (tag_list);
-
- fail_unless (gst_pad_send_event (pad, event),
- "Cannot send TAG event: Not handled.");
-
- gst_object_unref (pad);
-}
-
-static void
-poll_eos (GstElement * element)
-{
- GstBus *bus = gst_element_get_bus (element);
- GstMessage *message;
-
- message = gst_bus_poll (bus, GST_MESSAGE_EOS | GST_MESSAGE_TAG, GST_SECOND);
- fail_unless (message != NULL, "Could not poll for EOS message: Timed out");
- fail_unless (message->type == GST_MESSAGE_EOS,
- "Could not poll for eos message: got message of type %s instead",
- gst_message_type_get_name (message->type));
-
- gst_message_unref (message);
- gst_object_unref (bus);
-}
-
-/* This also polls for EOS since the TAG message comes right before the end of
- * streams. */
-
-static GstTagList *
-poll_tags (GstElement * element)
-{
- GstBus *bus = gst_element_get_bus (element);
- GstTagList *tag_list;
- GstMessage *message;
-
- message = gst_bus_poll (bus, GST_MESSAGE_TAG, GST_SECOND);
- fail_unless (message != NULL, "Could not poll for TAG message: Timed out");
-
- fail_unless (GST_MESSAGE_SRC (message) == GST_OBJECT (element));
-
- gst_message_parse_tag (message, &tag_list);
- gst_message_unref (message);
- gst_object_unref (bus);
-
- poll_eos (element);
-
- return tag_list;
-}
-
-#define MATCH_PEAK(p1, p2) ((p1 < p2 + 1e-6) && (p2 < p1 + 1e-6))
-#define MATCH_GAIN(g1, g2) ((g1 < g2 + 1e-13) && (g2 < g1 + 1e-13))
-
-static void
-fail_unless_track_gain (const GstTagList * tag_list, gdouble gain)
-{
- gdouble result;
-
- fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN, &result),
- "Tag list contains no track gain value");
- fail_unless (MATCH_GAIN (gain, result),
- "Track gain %+.2f does not match, expected %+.2f", result, gain);
-}
-
-static void
-fail_unless_track_peak (const GstTagList * tag_list, gdouble peak)
-{
- gdouble result;
-
- fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK, &result),
- "Tag list contains no track peak value");
- fail_unless (MATCH_PEAK (peak, result),
- "Track peak %f does not match, expected %f", result, peak);
-}
-
-static void
-fail_unless_album_gain (const GstTagList * tag_list, gdouble gain)
-{
- gdouble result;
-
- fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN, &result),
- "Tag list contains no album gain value");
- fail_unless (MATCH_GAIN (result, gain),
- "Album gain %+.2f does not match, expected %+.2f", result, gain);
-}
-
-static void
-fail_unless_album_peak (const GstTagList * tag_list, gdouble peak)
-{
- gdouble result;
-
- fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK, &result),
- "Tag list contains no album peak value");
- fail_unless (MATCH_PEAK (peak, result),
- "Album peak %f does not match, expected %f", result, peak);
-}
-
-static void
-fail_if_track_tags (const GstTagList * tag_list)
-{
- gdouble result;
-
- fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN, &result),
- "Tag list contains track gain value (but should not)");
- fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK, &result),
- "Tag list contains track peak value (but should not)");
-}
-
-static void
-fail_if_album_tags (const GstTagList * tag_list)
-{
- gdouble result;
-
- fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN, &result),
- "Tag list contains album gain value (but should not)");
- fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK, &result),
- "Tag list contains album peak value (but should not)");
-}
-
-static void
-fail_unless_num_tracks (GstElement * element, guint num_tracks)
-{
- guint current;
-
- g_object_get (element, "num-tracks", ¤t, NULL);
- fail_unless (current == num_tracks,
- "num-tracks property has incorrect value %u, expected %u",
- current, num_tracks);
-}
-
-/* Functions that create buffers with constant sample values, for peak
- * tests. */
-
-static GstBuffer *
-test_buffer_const_float_mono (gint sample_rate, gsize n_frames, gfloat value)
-{
- GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat));
- gfloat *data = (gfloat *) GST_BUFFER_DATA (buf);
- GstCaps *caps;
- gint i;
-
- for (i = n_frames; i--;)
- *data++ = value;
-
- caps = gst_caps_new_simple ("audio/x-raw-float",
- "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1,
- "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL);
- gst_buffer_set_caps (buf, caps);
- gst_caps_unref (caps);
-
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
-
- return buf;
-}
-
-static GstBuffer *
-test_buffer_const_float_stereo (gint sample_rate, gsize n_frames,
- gfloat value_l, gfloat value_r)
-{
- GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat) * 2);
- gfloat *data = (gfloat *) GST_BUFFER_DATA (buf);
- GstCaps *caps;
- gint i;
-
- for (i = n_frames; i--;) {
- *data++ = value_l;
- *data++ = value_r;
- }
-
- caps = gst_caps_new_simple ("audio/x-raw-float",
- "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2,
- "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL);
- gst_buffer_set_caps (buf, caps);
- gst_caps_unref (caps);
-
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
-
- return buf;
-}
-
-static GstBuffer *
-test_buffer_const_int16_mono (gint sample_rate, gint depth, gsize n_frames,
- gint16 value)
-{
- GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16));
- gint16 *data = (gint16 *) GST_BUFFER_DATA (buf);
- GstCaps *caps;
- gint i;
-
- for (i = n_frames; i--;)
- *data++ = value;
-
- caps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1,
- "endianness", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE,
- "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL);
- gst_buffer_set_caps (buf, caps);
- gst_caps_unref (caps);
-
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
-
- return buf;
-}
-
-static GstBuffer *
-test_buffer_const_int16_stereo (gint sample_rate, gint depth, gsize n_frames,
- gint16 value_l, gint16 value_r)
-{
- GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16) * 2);
- gint16 *data = (gint16 *) GST_BUFFER_DATA (buf);
- GstCaps *caps;
- gint i;
-
- for (i = n_frames; i--;) {
- *data++ = value_l;
- *data++ = value_r;
- }
-
- caps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2,
- "endianness", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE,
- "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL);
- gst_buffer_set_caps (buf, caps);
- gst_caps_unref (caps);
-
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
-
- return buf;
-}
-
-/* Functions that create data buffers containing square signal
- * waveforms. */
-
-static GstBuffer *
-test_buffer_square_float_mono (gint * accumulator, gint sample_rate,
- gsize n_frames, gfloat value)
-{
- GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat));
- gfloat *data = (gfloat *) GST_BUFFER_DATA (buf);
- GstCaps *caps;
- gint i;
-
- for (i = n_frames; i--;) {
- *accumulator += 1;
- *accumulator %= 96;
-
- if (*accumulator < 48)
- *data++ = value;
- else
- *data++ = -value;
- }
-
- caps = gst_caps_new_simple ("audio/x-raw-float",
- "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1,
- "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL);
- gst_buffer_set_caps (buf, caps);
- gst_caps_unref (caps);
-
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
-
- return buf;
-}
-
-static GstBuffer *
-test_buffer_square_float_stereo (gint * accumulator, gint sample_rate,
- gsize n_frames, gfloat value_l, gfloat value_r)
-{
- GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gfloat) * 2);
- gfloat *data = (gfloat *) GST_BUFFER_DATA (buf);
- GstCaps *caps;
- gint i;
-
- for (i = n_frames; i--;) {
- *accumulator += 1;
- *accumulator %= 96;
-
- if (*accumulator < 48) {
- *data++ = value_l;
- *data++ = value_r;
- } else {
- *data++ = -value_l;
- *data++ = -value_r;
- }
- }
-
- caps = gst_caps_new_simple ("audio/x-raw-float",
- "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2,
- "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL);
- gst_buffer_set_caps (buf, caps);
- gst_caps_unref (caps);
-
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
-
- return buf;
-}
-
-static GstBuffer *
-test_buffer_square_int16_mono (gint * accumulator, gint sample_rate,
- gint depth, gsize n_frames, gint16 value)
-{
- GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16));
- gint16 *data = (gint16 *) GST_BUFFER_DATA (buf);
- GstCaps *caps;
- gint i;
-
- for (i = n_frames; i--;) {
- *accumulator += 1;
- *accumulator %= 96;
-
- if (*accumulator < 48)
- *data++ = value;
- else
- *data++ = -MAX (value, -32767);
- }
-
- caps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 1,
- "endianness", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE,
- "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL);
- gst_buffer_set_caps (buf, caps);
- gst_caps_unref (caps);
-
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
-
- return buf;
-}
-
-static GstBuffer *
-test_buffer_square_int16_stereo (gint * accumulator, gint sample_rate,
- gint depth, gsize n_frames, gint16 value_l, gint16 value_r)
-{
- GstBuffer *buf = gst_buffer_new_and_alloc (n_frames * sizeof (gint16) * 2);
- gint16 *data = (gint16 *) GST_BUFFER_DATA (buf);
- GstCaps *caps;
- gint i;
-
- for (i = n_frames; i--;) {
- *accumulator += 1;
- *accumulator %= 96;
-
- if (*accumulator < 48) {
- *data++ = value_l;
- *data++ = value_r;
- } else {
- *data++ = -MAX (value_l, -32767);
- *data++ = -MAX (value_r, -32767);
- }
- }
-
- caps = gst_caps_new_simple ("audio/x-raw-int",
- "rate", G_TYPE_INT, sample_rate, "channels", G_TYPE_INT, 2,
- "endianness", G_TYPE_INT, G_BYTE_ORDER, "signed", G_TYPE_BOOLEAN, TRUE,
- "width", G_TYPE_INT, 16, "depth", G_TYPE_INT, depth, NULL);
- gst_buffer_set_caps (buf, caps);
- gst_caps_unref (caps);
-
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
-
- return buf;
-}
-
-static void
-push_buffer (GstBuffer * buf)
-{
- /* gst_pad_push steals a reference. */
- fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
-}
-
-/*** Start of the tests. ***/
-
-/* This test looks redundant, but early versions of the element
- * crashed when doing, well, nothing: */
-
-GST_START_TEST (test_no_buffer)
-{
- GstElement *element = setup_rganalysis ();
-
- set_playing_state (element);
- send_eos_event (element);
- poll_eos (element);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_no_buffer_album_1)
-{
- GstElement *element = setup_rganalysis ();
-
- set_playing_state (element);
-
- /* Single track: */
- send_eos_event (element);
- poll_eos (element);
-
- /* First album: */
- g_object_set (element, "num-tracks", 3, NULL);
-
- send_eos_event (element);
- poll_eos (element);
- fail_unless_num_tracks (element, 2);
-
- send_eos_event (element);
- poll_eos (element);
- fail_unless_num_tracks (element, 1);
-
- send_eos_event (element);
- poll_eos (element);
- fail_unless_num_tracks (element, 0);
-
- /* Second album: */
- g_object_set (element, "num-tracks", 2, NULL);
-
- send_eos_event (element);
- poll_eos (element);
- fail_unless_num_tracks (element, 1);
-
- send_eos_event (element);
- poll_eos (element);
- fail_unless_num_tracks (element, 0);
-
- /* Single track: */
- send_eos_event (element);
- poll_eos (element);
- fail_unless_num_tracks (element, 0);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_no_buffer_album_2)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- gint accumulator = 0;
- gint i;
-
- g_object_set (element, "num-tracks", 3, NULL);
- set_playing_state (element);
-
- /* No buffer for the first track. */
-
- send_eos_event (element);
- /* No tags should be posted, there was nothing to analyze: */
- poll_eos (element);
- fail_unless_num_tracks (element, 2);
-
- /* A test waveform with known gain result as second track: */
-
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_mono (&accumulator, 44100, 512,
- 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, -6.20);
- /* Album is not finished yet: */
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 1);
-
- /* No buffer for the last track. */
-
- send_eos_event (element);
-
- tag_list = poll_tags (element);
- fail_unless_album_peak (tag_list, 0.25);
- fail_unless_album_gain (tag_list, -6.20);
- /* No track tags should be posted, as there was no data for it: */
- fail_if_track_tags (tag_list);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 0);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_empty_buffers)
-{
- GstElement *element = setup_rganalysis ();
-
- set_playing_state (element);
-
- /* Single track: */
- push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0));
- send_eos_event (element);
- poll_eos (element);
-
- /* First album: */
- g_object_set (element, "num-tracks", 2, NULL);
-
- push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0));
- send_eos_event (element);
- poll_eos (element);
- fail_unless_num_tracks (element, 1);
-
- push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0));
- send_eos_event (element);
- poll_eos (element);
- fail_unless_num_tracks (element, 0);
-
- /* Second album, with a single track: */
- g_object_set (element, "num-tracks", 1, NULL);
- push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0));
- send_eos_event (element);
- poll_eos (element);
- fail_unless_num_tracks (element, 0);
-
- /* Single track: */
- push_buffer (test_buffer_const_float_stereo (44100, 0, 0.0, 0.0));
- send_eos_event (element);
- poll_eos (element);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_gap_buffers)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- GstBuffer *buf;
- gint accumulator = 0;
- gint i;
-
- set_playing_state (element);
-
- for (i = 0; i < 60; i++) {
- if (i % 3 == 0) {
- /* We are cheating here; the element cannot know that these GAP buffers
- * actually contain non-silence so it must skip them. */
- buf = test_buffer_square_float_mono (&accumulator, 44100, 512, 0.25);
- GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_GAP);
- push_buffer (buf);
-
- /* Verify that the base class does not lift the GAP flag: */
- fail_if (g_list_length (buffers) == 0);
- if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP))
- fail_unless (GST_BUFFER_FLAG_IS_SET (buffers->data,
- GST_BUFFER_FLAG_GAP));
- } else {
- push_buffer (test_buffer_const_float_mono (44100, 512, 0.0));
- }
- }
-
- send_eos_event (element);
- tag_list = poll_tags (element);
- /* We pushed faked GAP buffers with non-silence and non-GAP buffers with
- * silence, so the correct result is that the analysis only got silence: */
- fail_unless_track_peak (tag_list, 0.0);
- fail_unless_track_gain (tag_list, SILENCE_GAIN);
-
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-/* Tests for correctness of the peak values. */
-
-/* Float peak test. For stereo, one channel has the constant value of -1.369,
- * the other one 0.0. This tests many things: The result peak value should
- * occur on any channel. The peak is of course the absolute amplitude, so 1.369
- * should be the result. This will also detect if the code uses the absolute
- * value during the comparison. If it is buggy it will return 0.0 since 0.0 >
- * -1.369. Furthermore, this makes sure that there is no problem with headroom
- * (exceeding 0dBFS). In the wild you get float samples > 1.0 from stuff like
- * vorbis. */
-
-GST_START_TEST (test_peak_float)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
-
- set_playing_state (element);
- push_buffer (test_buffer_const_float_stereo (8000, 512, -1.369, 0.0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.369);
- gst_tag_list_free (tag_list);
-
- /* Swapped channels. */
- push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, -1.369));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.369);
- gst_tag_list_free (tag_list);
-
- /* Mono. */
- push_buffer (test_buffer_const_float_mono (8000, 512, -1.369));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.369);
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_peak_int16_16)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
-
- set_playing_state (element);
-
- /* Half amplitude. */
- push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 1 << 14, 0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
- /* Swapped channels. */
- push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, 1 << 14));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
- /* Mono. */
- push_buffer (test_buffer_const_int16_mono (8000, 16, 512, 1 << 14));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
- /* Half amplitude, negative variant. */
- push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -1 << 14, 0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
- /* Swapped channels. */
- push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -1 << 14));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
- /* Mono. */
- push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -1 << 14));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
-
- /* Now check for correct normalization of the peak value: Sample
- * values of this format range from -32768 to 32767. So for the
- * highest positive amplitude we do not reach 1.0, only for
- * -32768! */
-
- push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 32767, 0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 32767. / 32768.);
- gst_tag_list_free (tag_list);
-
- /* Swapped channels. */
- push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, 32767));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 32767. / 32768.);
- gst_tag_list_free (tag_list);
-
- /* Mono. */
- push_buffer (test_buffer_const_int16_mono (8000, 16, 512, 32767));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 32767. / 32768.);
- gst_tag_list_free (tag_list);
-
-
- /* Negative variant, reaching 1.0. */
- push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, -32768, 0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.0);
- gst_tag_list_free (tag_list);
-
- /* Swapped channels. */
- push_buffer (test_buffer_const_int16_stereo (8000, 16, 512, 0, -32768));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.0);
- gst_tag_list_free (tag_list);
-
- /* Mono. */
- push_buffer (test_buffer_const_int16_mono (8000, 16, 512, -32768));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.0);
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-/* Same as the test before, but with 8 bits (packed into 16 bits). */
-
-GST_START_TEST (test_peak_int16_8)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
-
- set_playing_state (element);
-
- /* Half amplitude. */
- push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 1 << 6, 0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
- /* Swapped channels. */
- push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, 1 << 6));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
- /* Mono. */
- push_buffer (test_buffer_const_int16_mono (8000, 8, 512, 1 << 6));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
-
- /* Half amplitude, negative variant. */
- push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, -1 << 6, 0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
- /* Swapped channels. */
- push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, -1 << 6));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
- /* Mono. */
- push_buffer (test_buffer_const_int16_mono (8000, 8, 512, -1 << 6));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
-
-
- /* Almost full amplitude (maximum positive value). */
- push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, (1 << 7) - 1, 0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.9921875);
- gst_tag_list_free (tag_list);
-
- /* Swapped channels. */
- push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, (1 << 7) - 1));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.9921875);
- gst_tag_list_free (tag_list);
-
- /* Mono. */
- push_buffer (test_buffer_const_int16_mono (8000, 8, 512, (1 << 7) - 1));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.9921875);
- gst_tag_list_free (tag_list);
-
-
- /* Full amplitude (maximum negative value). */
- push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, -1 << 7, 0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.0);
- gst_tag_list_free (tag_list);
-
- /* Swapped channels. */
- push_buffer (test_buffer_const_int16_stereo (8000, 8, 512, 0, -1 << 7));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.0);
- gst_tag_list_free (tag_list);
-
- /* Mono. */
- push_buffer (test_buffer_const_int16_mono (8000, 8, 512, -1 << 7));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.0);
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_peak_album)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
-
- g_object_set (element, "num-tracks", 2, NULL);
- set_playing_state (element);
-
- push_buffer (test_buffer_const_float_stereo (8000, 1024, 1.0, 0.0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.0);
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 1);
-
- push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.0, 0.5));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- fail_unless_album_peak (tag_list, 1.0);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 0);
-
- /* Try a second album: */
- g_object_set (element, "num-tracks", 3, NULL);
-
- push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.4, 0.4));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.4);
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 2);
-
- push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.45, 0.45));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.45);
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 1);
-
- push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.2, 0.2));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.2);
- fail_unless_album_peak (tag_list, 0.45);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 0);
-
- /* And now a single track, not in album mode (num-tracks is 0
- * now): */
- push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.1, 0.1));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.1);
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-/* Switching from track to album mode. */
-
-GST_START_TEST (test_peak_track_album)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
-
- set_playing_state (element);
-
- push_buffer (test_buffer_const_float_mono (8000, 1024, 1.0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.0);
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
-
- g_object_set (element, "num-tracks", 1, NULL);
- push_buffer (test_buffer_const_float_mono (8000, 1024, 0.5));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- fail_unless_album_peak (tag_list, 0.5);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 0);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-/* Disabling album processing before the end of the album. Probably a rare edge
- * case and applications should not rely on this to work. They need to send the
- * element to the READY state to clear up after an aborted album anyway since
- * they might need to process another album afterwards. */
-
-GST_START_TEST (test_peak_album_abort_to_track)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
-
- g_object_set (element, "num-tracks", 2, NULL);
- set_playing_state (element);
-
- push_buffer (test_buffer_const_float_stereo (8000, 1024, 1.0, 0.0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 1.0);
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 1);
-
- g_object_set (element, "num-tracks", 0, NULL);
-
- push_buffer (test_buffer_const_float_stereo (8000, 1024, 0.0, 0.5));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_gain_album)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- gint accumulator;
- gint i;
-
- g_object_set (element, "num-tracks", 3, NULL);
- set_playing_state (element);
-
- /* The three tracks are constructed such that if any of these is in fact
- * ignored for the album gain, the album gain will differ. */
-
- accumulator = 0;
- for (i = 8; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.75, 0.75));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.75);
- fail_unless_track_gain (tag_list, -15.70);
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
-
- accumulator = 0;
- for (i = 12; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.5, 0.5));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.5);
- fail_unless_track_gain (tag_list, -12.22);
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
-
- accumulator = 0;
- for (i = 180; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.25, 0.25));
- send_eos_event (element);
-
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, -6.20);
- fail_unless_album_peak (tag_list, 0.75);
- /* Strangely, wavegain reports -12.17 for the album, but the fixed
- * metaflac agrees to us. Could be a 32767 vs. 32768 issue. */
- fail_unless_album_gain (tag_list, -12.18);
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-/* Checks ensuring that the "forced" property works as advertised. */
-
-GST_START_TEST (test_forced)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- gint accumulator = 0;
- gint i;
-
- g_object_set (element, "forced", FALSE, NULL);
- set_playing_state (element);
-
- tag_list = gst_tag_list_new ();
- /* Provided values are totally arbitrary. */
- gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
- GST_TAG_TRACK_PEAK, 1.0, GST_TAG_TRACK_GAIN, 2.21, NULL);
- send_tag_event (element, tag_list);
-
- for (i = 20; i--;)
- push_buffer (test_buffer_const_float_stereo (44100, 512, 0.5, 0.5));
- send_eos_event (element);
- /* This fails if a tag message is generated: */
- poll_eos (element);
-
- /* Now back to a track without tags. */
-
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.25, 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, get_expected_gain (44100));
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-/* Sending track gain and peak in separate tag lists. */
-
-GST_START_TEST (test_forced_separate)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- gint accumulator = 0;
- gint i;
-
- g_object_set (element, "forced", FALSE, NULL);
- set_playing_state (element);
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, GST_TAG_TRACK_GAIN, 2.21,
- NULL);
- send_tag_event (element, tag_list);
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, GST_TAG_TRACK_PEAK, 1.0,
- NULL);
- send_tag_event (element, tag_list);
-
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.5, 0.5));
- send_eos_event (element);
- /* This fails if a tag message is generated: */
- poll_eos (element);
-
- /* Now a track without tags. */
-
- accumulator = 0;
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.25, 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, get_expected_gain (44100));
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-/* A TAG event is sent _after_ data has already been processed. In real
- * pipelines, this could happen if there is more than one rganalysis element (by
- * accident). While it would have analyzed all the data prior to receiving the
- * event, I expect it to not post its results if not forced. This test is
- * almost equivalent to test_forced. */
-
-GST_START_TEST (test_forced_after_data)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- gint accumulator = 0;
- gint i;
-
- g_object_set (element, "forced", FALSE, NULL);
- set_playing_state (element);
-
- for (i = 20; i--;)
- push_buffer (test_buffer_const_float_stereo (8000, 512, 0.5, 0.5));
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
- GST_TAG_TRACK_PEAK, 1.0, GST_TAG_TRACK_GAIN, 2.21, NULL);
- send_tag_event (element, tag_list);
-
- send_eos_event (element);
- poll_eos (element);
-
- /* Now back to a normal track, this one has no tags: */
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25,
- 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, get_expected_gain (8000));
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-/* Like test_forced, but *analyze* an album afterwards. The two tests following
- * this one check the *skipping* of albums. */
-
-GST_START_TEST (test_forced_album)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- gint accumulator;
- gint i;
-
- g_object_set (element, "forced", FALSE, NULL);
- set_playing_state (element);
-
- tag_list = gst_tag_list_new ();
- /* Provided values are totally arbitrary. */
- gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
- GST_TAG_TRACK_PEAK, 1.0, GST_TAG_TRACK_GAIN, 2.21, NULL);
- send_tag_event (element, tag_list);
-
- accumulator = 0;
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.5, 0.5));
- send_eos_event (element);
- /* This fails if a tag message is generated: */
- poll_eos (element);
-
- /* Now an album without tags. */
- g_object_set (element, "num-tracks", 2, NULL);
-
- accumulator = 0;
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.25, 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, get_expected_gain (44100));
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 1);
-
- accumulator = 0;
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.25, 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, get_expected_gain (44100));
- fail_unless_album_peak (tag_list, 0.25);
- fail_unless_album_gain (tag_list, get_expected_gain (44100));
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 0);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_forced_album_skip)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- gint accumulator = 0;
- gint i;
-
- g_object_set (element, "forced", FALSE, "num-tracks", 2, NULL);
- set_playing_state (element);
-
- tag_list = gst_tag_list_new ();
- /* Provided values are totally arbitrary. */
- gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
- GST_TAG_TRACK_PEAK, 0.75, GST_TAG_TRACK_GAIN, 2.21,
- GST_TAG_ALBUM_PEAK, 0.80, GST_TAG_ALBUM_GAIN, -0.11, NULL);
- send_tag_event (element, tag_list);
-
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25,
- 0.25));
- send_eos_event (element);
- poll_eos (element);
- fail_unless_num_tracks (element, 1);
-
- /* This track has no tags, but needs to be skipped anyways since we
- * are in album processing mode. */
- for (i = 20; i--;)
- push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, 0.0));
- send_eos_event (element);
- poll_eos (element);
- fail_unless_num_tracks (element, 0);
-
- /* Normal track after the album. Of course not to be skipped. */
- accumulator = 0;
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25,
- 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, get_expected_gain (8000));
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_forced_album_no_skip)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- gint accumulator = 0;
- gint i;
-
- g_object_set (element, "forced", FALSE, "num-tracks", 2, NULL);
- set_playing_state (element);
-
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25,
- 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, get_expected_gain (8000));
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 1);
-
- /* The second track has indeed full tags, but although being not forced, this
- * one has to be processed because album processing is on. */
- tag_list = gst_tag_list_new ();
- /* Provided values are totally arbitrary. */
- gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
- GST_TAG_TRACK_PEAK, 0.75, GST_TAG_TRACK_GAIN, 2.21,
- GST_TAG_ALBUM_PEAK, 0.80, GST_TAG_ALBUM_GAIN, -0.11, NULL);
- send_tag_event (element, tag_list);
- for (i = 20; i--;)
- push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, 0.0));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.0);
- fail_unless_track_gain (tag_list, SILENCE_GAIN);
- /* Second track was just silence so the album peak equals the first
- * track's peak. */
- fail_unless_album_peak (tag_list, 0.25);
- /* Statistical processing leads to the second track being
- * ignored for the gain (because it is so short): */
- fail_unless_album_gain (tag_list, get_expected_gain (8000));
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 0);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_forced_abort_album_no_skip)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- gint accumulator = 0;
- gint i;
-
- g_object_set (element, "forced", FALSE, "num-tracks", 2, NULL);
- set_playing_state (element);
-
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 8000, 512, 0.25,
- 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, get_expected_gain (8000));
- fail_if_album_tags (tag_list);
- gst_tag_list_free (tag_list);
- fail_unless_num_tracks (element, 1);
-
- /* Disabling album processing before end of album: */
- g_object_set (element, "num-tracks", 0, NULL);
-
- /* Processing a track that has to be skipped. */
- tag_list = gst_tag_list_new ();
- /* Provided values are totally arbitrary. */
- gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND,
- GST_TAG_TRACK_PEAK, 0.75, GST_TAG_TRACK_GAIN, 2.21,
- GST_TAG_ALBUM_PEAK, 0.80, GST_TAG_ALBUM_GAIN, -0.11, NULL);
- send_tag_event (element, tag_list);
- for (i = 20; i--;)
- push_buffer (test_buffer_const_float_stereo (8000, 512, 0.0, 0.0));
- send_eos_event (element);
- poll_eos (element);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_reference_level)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- gdouble ref_level;
- gint accumulator = 0;
- gint i;
-
- set_playing_state (element);
-
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.25, 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, get_expected_gain (44100));
- fail_if_album_tags (tag_list);
- fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
- &ref_level) && MATCH_GAIN (ref_level, 89.),
- "Incorrect reference level tag");
- gst_tag_list_free (tag_list);
-
- g_object_set (element, "reference-level", 83., "num-tracks", 2, NULL);
-
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.25, 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, get_expected_gain (44100) - 6.);
- fail_if_album_tags (tag_list);
- fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
- &ref_level) && MATCH_GAIN (ref_level, 83.),
- "Incorrect reference level tag");
- gst_tag_list_free (tag_list);
-
- accumulator = 0;
- for (i = 20; i--;)
- push_buffer (test_buffer_square_float_stereo (&accumulator, 44100, 512,
- 0.25, 0.25));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, get_expected_gain (44100) - 6.);
- fail_unless_album_peak (tag_list, 0.25);
- /* We provided the same waveform twice, with a reset separating
- * them. Therefore, the album gain matches the track gain. */
- fail_unless_album_gain (tag_list, get_expected_gain (44100) - 6.);
- fail_unless (gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL,
- &ref_level) && MATCH_GAIN (ref_level, 83.),
- "Incorrect reference level tag");
- gst_tag_list_free (tag_list);
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_all_formats)
-{
- GstElement *element = setup_rganalysis ();
- GstTagList *tag_list;
- gint accumulator = 0;
- gint i, j;
-
- set_playing_state (element);
- for (i = G_N_ELEMENTS (supported_rates); i--;) {
- accumulator = 0;
- for (j = 0; j < 4; j++)
- push_buffer (test_buffer_square_float_stereo (&accumulator,
- supported_rates[i].sample_rate, 512, 0.25, 0.25));
- for (j = 0; j < 3; j++)
- push_buffer (test_buffer_square_float_mono (&accumulator,
- supported_rates[i].sample_rate, 512, 0.25));
- for (j = 0; j < 4; j++)
- push_buffer (test_buffer_square_int16_stereo (&accumulator,
- supported_rates[i].sample_rate, 16, 512, 1 << 13, 1 << 13));
- for (j = 0; j < 3; j++)
- push_buffer (test_buffer_square_int16_mono (&accumulator,
- supported_rates[i].sample_rate, 16, 512, 1 << 13));
- for (j = 0; j < 3; j++)
- push_buffer (test_buffer_square_int16_stereo (&accumulator,
- supported_rates[i].sample_rate, 8, 512, 1 << 5, 1 << 5));
- for (j = 0; j < 3; j++)
- push_buffer (test_buffer_square_int16_mono (&accumulator,
- supported_rates[i].sample_rate, 8, 512, 1 << 5));
- send_eos_event (element);
- tag_list = poll_tags (element);
- fail_unless_track_peak (tag_list, 0.25);
- fail_unless_track_gain (tag_list, supported_rates[i].gain);
- gst_tag_list_free (tag_list);
- }
-
- cleanup_rganalysis (element);
-}
-
-GST_END_TEST;
-
-/* Checks ensuring all advertised supported sample rates are really
- * accepted, for integer and float, mono and stereo. This also
- * verifies that the correct gain is computed for all formats (except
- * odd bit depths). */
-
-#define MAKE_GAIN_TEST_FLOAT_MONO(sample_rate) \
- GST_START_TEST (test_gain_float_mono_##sample_rate) \
-{ \
- GstElement *element = setup_rganalysis (); \
- GstTagList *tag_list; \
- gint accumulator = 0; \
- gint i; \
- \
- set_playing_state (element); \
- \
- for (i = 0; i < 20; i++) \
- push_buffer (test_buffer_square_float_mono (&accumulator, \
- sample_rate, 512, 0.25)); \
- send_eos_event (element); \
- tag_list = poll_tags (element); \
- fail_unless_track_peak (tag_list, 0.25); \
- fail_unless_track_gain (tag_list, \
- get_expected_gain (sample_rate)); \
- gst_tag_list_free (tag_list); \
- \
- cleanup_rganalysis (element); \
-} \
- \
-GST_END_TEST;
-
-#define MAKE_GAIN_TEST_FLOAT_STEREO(sample_rate) \
- GST_START_TEST (test_gain_float_stereo_##sample_rate) \
-{ \
- GstElement *element = setup_rganalysis (); \
- GstTagList *tag_list; \
- gint accumulator = 0; \
- gint i; \
- \
- set_playing_state (element); \
- \
- for (i = 0; i < 20; i++) \
- push_buffer (test_buffer_square_float_stereo (&accumulator, \
- sample_rate, 512, 0.25, 0.25)); \
- send_eos_event (element); \
- tag_list = poll_tags (element); \
- fail_unless_track_peak (tag_list, 0.25); \
- fail_unless_track_gain (tag_list, \
- get_expected_gain (sample_rate)); \
- gst_tag_list_free (tag_list); \
- \
- cleanup_rganalysis (element); \
-} \
- \
-GST_END_TEST;
-
-#define MAKE_GAIN_TEST_INT16_MONO(sample_rate, depth) \
- GST_START_TEST (test_gain_int16_##depth##_mono_##sample_rate) \
-{ \
- GstElement *element = setup_rganalysis (); \
- GstTagList *tag_list; \
- gint accumulator = 0; \
- gint i; \
- \
- set_playing_state (element); \
- \
- for (i = 0; i < 20; i++) \
- push_buffer (test_buffer_square_int16_mono (&accumulator, \
- sample_rate, depth, 512, 1 << (13 + depth - 16))); \
- \
- send_eos_event (element); \
- tag_list = poll_tags (element); \
- fail_unless_track_peak (tag_list, 0.25); \
- fail_unless_track_gain (tag_list, \
- get_expected_gain (sample_rate)); \
- gst_tag_list_free (tag_list); \
- \
- cleanup_rganalysis (element); \
-} \
- \
-GST_END_TEST;
-
-#define MAKE_GAIN_TEST_INT16_STEREO(sample_rate, depth) \
- GST_START_TEST (test_gain_int16_##depth##_stereo_##sample_rate) \
-{ \
- GstElement *element = setup_rganalysis (); \
- GstTagList *tag_list; \
- gint accumulator = 0; \
- gint i; \
- \
- set_playing_state (element); \
- \
- for (i = 0; i < 20; i++) \
- push_buffer (test_buffer_square_int16_stereo (&accumulator, \
- sample_rate, depth, 512, 1 << (13 + depth - 16), \
- 1 << (13 + depth - 16))); \
- send_eos_event (element); \
- tag_list = poll_tags (element); \
- fail_unless_track_peak (tag_list, 0.25); \
- fail_unless_track_gain (tag_list, \
- get_expected_gain (sample_rate)); \
- gst_tag_list_free (tag_list); \
- \
- cleanup_rganalysis (element); \
-} \
- \
-GST_END_TEST;
-
-MAKE_GAIN_TEST_FLOAT_MONO (8000);
-MAKE_GAIN_TEST_FLOAT_MONO (11025);
-MAKE_GAIN_TEST_FLOAT_MONO (12000);
-MAKE_GAIN_TEST_FLOAT_MONO (16000);
-MAKE_GAIN_TEST_FLOAT_MONO (22050);
-MAKE_GAIN_TEST_FLOAT_MONO (24000);
-MAKE_GAIN_TEST_FLOAT_MONO (32000);
-MAKE_GAIN_TEST_FLOAT_MONO (44100);
-MAKE_GAIN_TEST_FLOAT_MONO (48000);
-
-MAKE_GAIN_TEST_FLOAT_STEREO (8000);
-MAKE_GAIN_TEST_FLOAT_STEREO (11025);
-MAKE_GAIN_TEST_FLOAT_STEREO (12000);
-MAKE_GAIN_TEST_FLOAT_STEREO (16000);
-MAKE_GAIN_TEST_FLOAT_STEREO (22050);
-MAKE_GAIN_TEST_FLOAT_STEREO (24000);
-MAKE_GAIN_TEST_FLOAT_STEREO (32000);
-MAKE_GAIN_TEST_FLOAT_STEREO (44100);
-MAKE_GAIN_TEST_FLOAT_STEREO (48000);
-
-MAKE_GAIN_TEST_INT16_MONO (8000, 16);
-MAKE_GAIN_TEST_INT16_MONO (11025, 16);
-MAKE_GAIN_TEST_INT16_MONO (12000, 16);
-MAKE_GAIN_TEST_INT16_MONO (16000, 16);
-MAKE_GAIN_TEST_INT16_MONO (22050, 16);
-MAKE_GAIN_TEST_INT16_MONO (24000, 16);
-MAKE_GAIN_TEST_INT16_MONO (32000, 16);
-MAKE_GAIN_TEST_INT16_MONO (44100, 16);
-MAKE_GAIN_TEST_INT16_MONO (48000, 16);
-
-MAKE_GAIN_TEST_INT16_STEREO (8000, 16);
-MAKE_GAIN_TEST_INT16_STEREO (11025, 16);
-MAKE_GAIN_TEST_INT16_STEREO (12000, 16);
-MAKE_GAIN_TEST_INT16_STEREO (16000, 16);
-MAKE_GAIN_TEST_INT16_STEREO (22050, 16);
-MAKE_GAIN_TEST_INT16_STEREO (24000, 16);
-MAKE_GAIN_TEST_INT16_STEREO (32000, 16);
-MAKE_GAIN_TEST_INT16_STEREO (44100, 16);
-MAKE_GAIN_TEST_INT16_STEREO (48000, 16);
-
-MAKE_GAIN_TEST_INT16_MONO (8000, 8);
-MAKE_GAIN_TEST_INT16_MONO (11025, 8);
-MAKE_GAIN_TEST_INT16_MONO (12000, 8);
-MAKE_GAIN_TEST_INT16_MONO (16000, 8);
-MAKE_GAIN_TEST_INT16_MONO (22050, 8);
-MAKE_GAIN_TEST_INT16_MONO (24000, 8);
-MAKE_GAIN_TEST_INT16_MONO (32000, 8);
-MAKE_GAIN_TEST_INT16_MONO (44100, 8);
-MAKE_GAIN_TEST_INT16_MONO (48000, 8);
-
-MAKE_GAIN_TEST_INT16_STEREO (8000, 8);
-MAKE_GAIN_TEST_INT16_STEREO (11025, 8);
-MAKE_GAIN_TEST_INT16_STEREO (12000, 8);
-MAKE_GAIN_TEST_INT16_STEREO (16000, 8);
-MAKE_GAIN_TEST_INT16_STEREO (22050, 8);
-MAKE_GAIN_TEST_INT16_STEREO (24000, 8);
-MAKE_GAIN_TEST_INT16_STEREO (32000, 8);
-MAKE_GAIN_TEST_INT16_STEREO (44100, 8);
-MAKE_GAIN_TEST_INT16_STEREO (48000, 8);
-
-Suite *
-rganalysis_suite (void)
-{
- Suite *s = suite_create ("rganalysis");
- TCase *tc_chain = tcase_create ("general");
-
- suite_add_tcase (s, tc_chain);
-
- tcase_add_test (tc_chain, test_no_buffer);
- tcase_add_test (tc_chain, test_no_buffer_album_1);
- tcase_add_test (tc_chain, test_no_buffer_album_2);
- tcase_add_test (tc_chain, test_empty_buffers);
- tcase_add_test (tc_chain, test_gap_buffers);
-
- tcase_add_test (tc_chain, test_peak_float);
- tcase_add_test (tc_chain, test_peak_int16_16);
- tcase_add_test (tc_chain, test_peak_int16_8);
-
- tcase_add_test (tc_chain, test_peak_album);
- tcase_add_test (tc_chain, test_peak_track_album);
- tcase_add_test (tc_chain, test_peak_album_abort_to_track);
-
- tcase_add_test (tc_chain, test_gain_album);
-
- tcase_add_test (tc_chain, test_forced);
- tcase_add_test (tc_chain, test_forced_separate);
- tcase_add_test (tc_chain, test_forced_after_data);
- tcase_add_test (tc_chain, test_forced_album);
- tcase_add_test (tc_chain, test_forced_album_skip);
- tcase_add_test (tc_chain, test_forced_album_no_skip);
- tcase_add_test (tc_chain, test_forced_abort_album_no_skip);
-
- tcase_add_test (tc_chain, test_reference_level);
-
- tcase_add_test (tc_chain, test_all_formats);
-
- tcase_add_test (tc_chain, test_gain_float_mono_8000);
- tcase_add_test (tc_chain, test_gain_float_mono_11025);
- tcase_add_test (tc_chain, test_gain_float_mono_12000);
- tcase_add_test (tc_chain, test_gain_float_mono_16000);
- tcase_add_test (tc_chain, test_gain_float_mono_22050);
- tcase_add_test (tc_chain, test_gain_float_mono_24000);
- tcase_add_test (tc_chain, test_gain_float_mono_32000);
- tcase_add_test (tc_chain, test_gain_float_mono_44100);
- tcase_add_test (tc_chain, test_gain_float_mono_48000);
-
- tcase_add_test (tc_chain, test_gain_float_stereo_8000);
- tcase_add_test (tc_chain, test_gain_float_stereo_11025);
- tcase_add_test (tc_chain, test_gain_float_stereo_12000);
- tcase_add_test (tc_chain, test_gain_float_stereo_16000);
- tcase_add_test (tc_chain, test_gain_float_stereo_22050);
- tcase_add_test (tc_chain, test_gain_float_stereo_24000);
- tcase_add_test (tc_chain, test_gain_float_stereo_32000);
- tcase_add_test (tc_chain, test_gain_float_stereo_44100);
- tcase_add_test (tc_chain, test_gain_float_stereo_48000);
-
- tcase_add_test (tc_chain, test_gain_int16_16_mono_8000);
- tcase_add_test (tc_chain, test_gain_int16_16_mono_11025);
- tcase_add_test (tc_chain, test_gain_int16_16_mono_12000);
- tcase_add_test (tc_chain, test_gain_int16_16_mono_16000);
- tcase_add_test (tc_chain, test_gain_int16_16_mono_22050);
- tcase_add_test (tc_chain, test_gain_int16_16_mono_24000);
- tcase_add_test (tc_chain, test_gain_int16_16_mono_32000);
- tcase_add_test (tc_chain, test_gain_int16_16_mono_44100);
- tcase_add_test (tc_chain, test_gain_int16_16_mono_48000);
-
- tcase_add_test (tc_chain, test_gain_int16_16_stereo_8000);
- tcase_add_test (tc_chain, test_gain_int16_16_stereo_11025);
- tcase_add_test (tc_chain, test_gain_int16_16_stereo_12000);
- tcase_add_test (tc_chain, test_gain_int16_16_stereo_16000);
- tcase_add_test (tc_chain, test_gain_int16_16_stereo_22050);
- tcase_add_test (tc_chain, test_gain_int16_16_stereo_24000);
- tcase_add_test (tc_chain, test_gain_int16_16_stereo_32000);
- tcase_add_test (tc_chain, test_gain_int16_16_stereo_44100);
- tcase_add_test (tc_chain, test_gain_int16_16_stereo_48000);
-
- tcase_add_test (tc_chain, test_gain_int16_8_mono_8000);
- tcase_add_test (tc_chain, test_gain_int16_8_mono_11025);
- tcase_add_test (tc_chain, test_gain_int16_8_mono_12000);
- tcase_add_test (tc_chain, test_gain_int16_8_mono_16000);
- tcase_add_test (tc_chain, test_gain_int16_8_mono_22050);
- tcase_add_test (tc_chain, test_gain_int16_8_mono_24000);
- tcase_add_test (tc_chain, test_gain_int16_8_mono_32000);
- tcase_add_test (tc_chain, test_gain_int16_8_mono_44100);
- tcase_add_test (tc_chain, test_gain_int16_8_mono_48000);
-
- tcase_add_test (tc_chain, test_gain_int16_8_stereo_8000);
- tcase_add_test (tc_chain, test_gain_int16_8_stereo_11025);
- tcase_add_test (tc_chain, test_gain_int16_8_stereo_12000);
- tcase_add_test (tc_chain, test_gain_int16_8_stereo_16000);
- tcase_add_test (tc_chain, test_gain_int16_8_stereo_22050);
- tcase_add_test (tc_chain, test_gain_int16_8_stereo_24000);
- tcase_add_test (tc_chain, test_gain_int16_8_stereo_32000);
- tcase_add_test (tc_chain, test_gain_int16_8_stereo_44100);
- tcase_add_test (tc_chain, test_gain_int16_8_stereo_48000);
-
- return s;
-}
-
-int
-main (int argc, char **argv)
-{
- gint nf;
-
- Suite *s = rganalysis_suite ();
- SRunner *sr = srunner_create (s);
-
- gst_check_init (&argc, &argv);
-
- srunner_run_all (sr, CK_ENV);
- nf = srunner_ntests_failed (sr);
- srunner_free (sr);
-
- return nf;
-}
+++ /dev/null
-/* GStreamer ReplayGain limiter
- *
- * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
- *
- * rglimiter.c: Unit test for the rglimiter element
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-#include <gst/check/gstcheck.h>
-
-#include <math.h>
-
-GList *buffers = NULL;
-
-/* For ease of programming we use globals to keep refs for our floating
- * src and sink pads we create; otherwise we always have to do get_pad,
- * get_peer, and then remove references in every test function */
-static GstPad *mysrcpad, *mysinkpad;
-
-#define RG_LIMITER_CAPS_TEMPLATE_STRING \
- "audio/x-raw-float, " \
- "width = (int) 32, " \
- "endianness = (int) BYTE_ORDER, " \
- "channels = (int) [ 1, MAX ], " \
- "rate = (int) [ 1, MAX ]"
-
-static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS (RG_LIMITER_CAPS_TEMPLATE_STRING)
- );
-static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS (RG_LIMITER_CAPS_TEMPLATE_STRING)
- );
-
-GstElement *
-setup_rglimiter ()
-{
- GstElement *element;
-
- GST_DEBUG ("setup_rglimiter");
- element = gst_check_setup_element ("rglimiter");
- mysrcpad = gst_check_setup_src_pad (element, &srctemplate, NULL);
- mysinkpad = gst_check_setup_sink_pad (element, &sinktemplate, NULL);
- gst_pad_set_active (mysrcpad, TRUE);
- gst_pad_set_active (mysinkpad, TRUE);
-
- return element;
-}
-
-void
-cleanup_rglimiter (GstElement * element)
-{
- GST_DEBUG ("cleanup_rglimiter");
-
- g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
- g_list_free (buffers);
- buffers = NULL;
-
- gst_check_teardown_src_pad (element);
- gst_check_teardown_sink_pad (element);
- gst_check_teardown_element (element);
-}
-
-static void
-set_playing_state (GstElement * element)
-{
- fail_unless (gst_element_set_state (element,
- GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
- "Could not set state to PLAYING");
-}
-
-static const gfloat test_input[] = {
- -2.0, -1.0, -0.75, -0.5, -0.25, 0.0, 0.25, 0.5, 0.75, 1.0, 2.0
-};
-static const gfloat test_output[] = {
- -0.99752737684336523, /* -2.0 */
- -0.88079707797788243, /* -1.0 */
- -0.7310585786300049, /* -0.75 */
- -0.5, -0.25, 0.0, 0.25, 0.5,
- 0.7310585786300049, /* 0.75 */
- 0.88079707797788243, /* 1.0 */
- 0.99752737684336523, /* 2.0 */
-};
-
-static GstBuffer *
-create_test_buffer ()
-{
- GstBuffer *buf = gst_buffer_new_and_alloc (sizeof (test_input));
- GstCaps *caps;
-
- memcpy (GST_BUFFER_DATA (buf), test_input, sizeof (test_input));
-
- caps = gst_caps_new_simple ("audio/x-raw-float",
- "rate", G_TYPE_INT, 44100, "channels", G_TYPE_INT, 1,
- "endianness", G_TYPE_INT, G_BYTE_ORDER, "width", G_TYPE_INT, 32, NULL);
- gst_buffer_set_caps (buf, caps);
- gst_caps_unref (caps);
-
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
-
- return buf;
-}
-
-static void
-verify_test_buffer (GstBuffer * buf)
-{
- gfloat *output = (gfloat *) GST_BUFFER_DATA (buf);
- gint i;
-
- fail_unless (GST_BUFFER_SIZE (buf) == sizeof (test_output));
- for (i = 0; i < G_N_ELEMENTS (test_input); i++)
- fail_unless (ABS (output[i] - test_output[i]) < 1.e-6,
- "Incorrect output value %.6f for input %.2f, expected %.6f",
- output[i], test_input[i], test_output[i]);
-}
-
-/* Start of tests. */
-
-GST_START_TEST (test_no_buffer)
-{
- GstElement *element = setup_rglimiter ();
-
- set_playing_state (element);
-
- cleanup_rglimiter (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_disabled)
-{
- GstElement *element = setup_rglimiter ();
- GstBuffer *buf, *out_buf;
-
- g_object_set (element, "enabled", FALSE, NULL);
- set_playing_state (element);
-
- buf = create_test_buffer ();
- fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
- fail_unless (g_list_length (buffers) == 1);
- out_buf = buffers->data;
- fail_if (out_buf == NULL);
- buffers = g_list_remove (buffers, out_buf);
- ASSERT_BUFFER_REFCOUNT (out_buf, "out_buf", 1);
- fail_unless (buf == out_buf);
- gst_buffer_unref (out_buf);
-
- cleanup_rglimiter (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_limiting)
-{
- GstElement *element = setup_rglimiter ();
- GstBuffer *buf, *out_buf;
-
- set_playing_state (element);
-
- /* Mutable variant. */
- buf = create_test_buffer ();
- fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
- fail_unless (g_list_length (buffers) == 1);
- out_buf = buffers->data;
- fail_if (out_buf == NULL);
- ASSERT_BUFFER_REFCOUNT (out_buf, "out_buf", 1);
- verify_test_buffer (out_buf);
-
- /* Immutable variant. */
- buf = create_test_buffer ();
- /* Extra ref: */
- gst_buffer_ref (buf);
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 2);
- fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
- fail_unless (g_list_length (buffers) == 2);
- out_buf = g_list_last (buffers)->data;
- fail_if (out_buf == NULL);
- ASSERT_BUFFER_REFCOUNT (out_buf, "out_buf", 1);
- fail_unless (buf != out_buf);
- /* Drop our extra ref: */
- gst_buffer_unref (buf);
- verify_test_buffer (out_buf);
-
- cleanup_rglimiter (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_gap)
-{
- GstElement *element = setup_rglimiter ();
- GstBuffer *buf, *out_buf;
-
- set_playing_state (element);
-
- buf = create_test_buffer ();
- GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_GAP);
- fail_unless (gst_pad_push (mysrcpad, buf) == GST_FLOW_OK);
- fail_unless (g_list_length (buffers) == 1);
- out_buf = buffers->data;
- fail_if (out_buf == NULL);
- ASSERT_BUFFER_REFCOUNT (out_buf, "out_buf", 1);
-
- /* Verify that the baseclass does not lift the GAP flag: */
- fail_unless (GST_BUFFER_FLAG_IS_SET (out_buf, GST_BUFFER_FLAG_GAP));
-
- g_assert (GST_BUFFER_SIZE (out_buf) == GST_BUFFER_SIZE (buf));
- /* We cheated by passing an input buffer with non-silence that has the GAP
- * flag set. The element cannot know that however and must have skipped
- * adjusting the buffer because of the flag, which we can easily verify: */
- fail_if (memcmp (GST_BUFFER_DATA (out_buf),
- GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (out_buf)) != 0);
-
- cleanup_rglimiter (element);
-}
-
-GST_END_TEST;
-
-Suite *
-rglimiter_suite (void)
-{
- Suite *s = suite_create ("rglimiter");
- TCase *tc_chain = tcase_create ("general");
-
- suite_add_tcase (s, tc_chain);
-
- tcase_add_test (tc_chain, test_no_buffer);
- tcase_add_test (tc_chain, test_disabled);
- tcase_add_test (tc_chain, test_limiting);
- tcase_add_test (tc_chain, test_gap);
-
- return s;
-}
-
-int
-main (int argc, char **argv)
-{
- gint nf;
-
- Suite *s = rglimiter_suite ();
- SRunner *sr = srunner_create (s);
-
- gst_check_init (&argc, &argv);
-
- srunner_run_all (sr, CK_ENV);
- nf = srunner_ntests_failed (sr);
- srunner_free (sr);
-
- return nf;
-}
+++ /dev/null
-/* GStreamer ReplayGain volume adjustment
- *
- * Copyright (C) 2007 Rene Stadler <mail@renestadler.de>
- *
- * rgvolume.c: Unit test for the rgvolume element
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public License
- * as published by the Free Software Foundation; either version 2.1 of
- * the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful, but
- * WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA
- * 02110-1301 USA
- */
-
-#include <gst/check/gstcheck.h>
-
-#include <math.h>
-
-GList *buffers = NULL;
-GList *events = NULL;
-
-/* For ease of programming we use globals to keep refs for our floating src and
- * sink pads we create; otherwise we always have to do get_pad, get_peer, and
- * then remove references in every test function */
-static GstPad *mysrcpad, *mysinkpad;
-
-#define RG_VOLUME_CAPS_TEMPLATE_STRING \
- "audio/x-raw-float, " \
- "width = (int) 32, " \
- "endianness = (int) BYTE_ORDER, " \
- "channels = (int) [ 1, MAX ], " \
- "rate = (int) [ 1, MAX ]"
-
-static GstStaticPadTemplate sinktemplate = GST_STATIC_PAD_TEMPLATE ("sink",
- GST_PAD_SINK,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS (RG_VOLUME_CAPS_TEMPLATE_STRING)
- );
-static GstStaticPadTemplate srctemplate = GST_STATIC_PAD_TEMPLATE ("src",
- GST_PAD_SRC,
- GST_PAD_ALWAYS,
- GST_STATIC_CAPS (RG_VOLUME_CAPS_TEMPLATE_STRING)
- );
-
-/* gstcheck sets up a chain function that appends buffers to a global list.
- * This is our equivalent of that for event handling. */
-static gboolean
-event_func (GstPad * pad, GstEvent * event)
-{
- events = g_list_append (events, event);
-
- return TRUE;
-}
-
-GstElement *
-setup_rgvolume ()
-{
- GstElement *element;
-
- GST_DEBUG ("setup_rgvolume");
- element = gst_check_setup_element ("rgvolume");
- mysrcpad = gst_check_setup_src_pad (element, &srctemplate, NULL);
- mysinkpad = gst_check_setup_sink_pad (element, &sinktemplate, NULL);
-
- /* Capture events, to test tag filtering behavior: */
- gst_pad_set_event_function (mysinkpad, event_func);
-
- gst_pad_set_active (mysrcpad, TRUE);
- gst_pad_set_active (mysinkpad, TRUE);
-
- return element;
-}
-
-void
-cleanup_rgvolume (GstElement * element)
-{
- GST_DEBUG ("cleanup_rgvolume");
-
- g_list_foreach (buffers, (GFunc) gst_mini_object_unref, NULL);
- g_list_free (buffers);
- buffers = NULL;
-
- g_list_foreach (events, (GFunc) gst_mini_object_unref, NULL);
- g_list_free (events);
- events = NULL;
-
- gst_pad_set_active (mysrcpad, FALSE);
- gst_pad_set_active (mysinkpad, FALSE);
- gst_check_teardown_src_pad (element);
- gst_check_teardown_sink_pad (element);
- gst_check_teardown_element (element);
-}
-
-static void
-set_playing_state (GstElement * element)
-{
- fail_unless (gst_element_set_state (element,
- GST_STATE_PLAYING) == GST_STATE_CHANGE_SUCCESS,
- "Could not set state to PLAYING");
-}
-
-static void
-set_null_state (GstElement * element)
-{
- fail_unless (gst_element_set_state (element,
- GST_STATE_NULL) == GST_STATE_CHANGE_SUCCESS,
- "Could not set state to NULL");
-}
-
-static void
-send_eos_event (GstElement * element)
-{
- GstEvent *event = gst_event_new_eos ();
-
- fail_unless (g_list_length (events) == 0);
- fail_unless (gst_pad_push_event (mysrcpad, event),
- "Pushing EOS event failed");
- fail_unless (g_list_length (events) == 1);
- fail_unless (events->data == event);
- gst_mini_object_unref ((GstMiniObject *) events->data);
- events = g_list_remove (events, event);
-}
-
-static GstEvent *
-send_tag_event (GstElement * element, GstEvent * event)
-{
- g_return_val_if_fail (event->type == GST_EVENT_TAG, NULL);
-
- fail_unless (g_list_length (events) == 0);
- fail_unless (gst_pad_push_event (mysrcpad, event),
- "Pushing tag event failed");
-
- if (g_list_length (events) == 0) {
- /* Event got filtered out. */
- event = NULL;
- } else {
- GstTagList *tag_list;
- gdouble dummy;
-
- event = events->data;
- events = g_list_remove (events, event);
-
- fail_unless (event->type == GST_EVENT_TAG);
- gst_event_parse_tag (event, &tag_list);
-
- /* The element is supposed to filter out ReplayGain related tags. */
- fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN, &dummy),
- "tag event still contains track gain tag");
- fail_if (gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK, &dummy),
- "tag event still contains track peak tag");
- fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN, &dummy),
- "tag event still contains album gain tag");
- fail_if (gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK, &dummy),
- "tag event still contains album peak tag");
- }
-
- return event;
-}
-
-static GstBuffer *
-test_buffer_new (gfloat value)
-{
- GstBuffer *buf;
- GstCaps *caps;
- gfloat *data;
- gint i;
-
- buf = gst_buffer_new_and_alloc (8 * sizeof (gfloat));
- data = (gfloat *) GST_BUFFER_DATA (buf);
- for (i = 0; i < 8; i++)
- data[i] = value;
-
- caps = gst_caps_from_string ("audio/x-raw-float, "
- "rate = 8000, channels = 1, endianness = BYTE_ORDER, width = 32");
- gst_buffer_set_caps (buf, caps);
- gst_caps_unref (caps);
-
- ASSERT_BUFFER_REFCOUNT (buf, "buf", 1);
-
- return buf;
-}
-
-#define MATCH_GAIN(g1, g2) ((g1 < g2 + 1e-6) && (g2 < g1 + 1e-6))
-
-static void
-fail_unless_target_gain (GstElement * element, gdouble expected_gain)
-{
- gdouble prop_gain;
-
- g_object_get (element, "target-gain", &prop_gain, NULL);
-
- fail_unless (MATCH_GAIN (prop_gain, expected_gain),
- "Target gain is %.2f dB, expected %.2f dB", prop_gain, expected_gain);
-}
-
-static void
-fail_unless_result_gain (GstElement * element, gdouble expected_gain)
-{
- GstBuffer *input_buf, *output_buf;
- gfloat input_sample, output_sample;
- gdouble gain, prop_gain;
- gboolean is_passthrough, expect_passthrough;
- gint i;
-
- fail_unless (g_list_length (buffers) == 0);
-
- input_sample = 1.0;
- input_buf = test_buffer_new (input_sample);
-
- /* We keep an extra reference to detect passthrough mode. */
- gst_buffer_ref (input_buf);
- /* Pushing steals a reference. */
- fail_unless (gst_pad_push (mysrcpad, input_buf) == GST_FLOW_OK);
- gst_buffer_unref (input_buf);
-
- /* The output buffer ends up on the global buffer list. */
- fail_unless (g_list_length (buffers) == 1);
- output_buf = buffers->data;
- fail_if (output_buf == NULL);
-
- buffers = g_list_remove (buffers, output_buf);
- ASSERT_BUFFER_REFCOUNT (output_buf, "output_buf", 1);
- fail_unless_equals_int (GST_BUFFER_SIZE (output_buf), 8 * sizeof (gfloat));
-
- output_sample = *((gfloat *) GST_BUFFER_DATA (output_buf));
-
- fail_if (output_sample == 0.0, "First output sample is zero");
- for (i = 1; i < 8; i++) {
- gfloat output = ((gfloat *) GST_BUFFER_DATA (output_buf))[i];
-
- fail_unless (output_sample == output, "Output samples not uniform");
- };
-
- gain = 20. * log10 (output_sample / input_sample);
- fail_unless (MATCH_GAIN (gain, expected_gain),
- "Applied gain is %.2f dB, expected %.2f dB", gain, expected_gain);
- g_object_get (element, "result-gain", &prop_gain, NULL);
- fail_unless (MATCH_GAIN (prop_gain, expected_gain),
- "Result gain is %.2f dB, expected %.2f dB", prop_gain, expected_gain);
-
- is_passthrough = (output_buf == input_buf);
- expect_passthrough = MATCH_GAIN (expected_gain, +0.00);
- fail_unless (is_passthrough == expect_passthrough,
- expect_passthrough
- ? "Expected operation in passthrough mode"
- : "Incorrect passthrough behaviour");
-
- gst_buffer_unref (output_buf);
-}
-
-static void
-fail_unless_gain (GstElement * element, gdouble expected_gain)
-{
- fail_unless_target_gain (element, expected_gain);
- fail_unless_result_gain (element, expected_gain);
-}
-
-/* Start of tests. */
-
-GST_START_TEST (test_no_buffer)
-{
- GstElement *element = setup_rgvolume ();
-
- set_playing_state (element);
- set_null_state (element);
- set_playing_state (element);
- send_eos_event (element);
-
- cleanup_rgvolume (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_events)
-{
- GstElement *element = setup_rgvolume ();
- GstEvent *event;
- GstEvent *new_event;
- GstTagList *tag_list;
- gchar *artist;
-
- set_playing_state (element);
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_TRACK_GAIN, +4.95, GST_TAG_TRACK_PEAK, 0.59463,
- GST_TAG_ALBUM_GAIN, -1.54, GST_TAG_ALBUM_PEAK, 0.693415,
- GST_TAG_ARTIST, "Foobar", NULL);
- event = gst_event_new_tag (tag_list);
- new_event = send_tag_event (element, event);
- /* Expect the element to modify the writable event. */
- fail_unless (event == new_event, "Writable tag event not reused");
- gst_event_parse_tag (new_event, &tag_list);
- fail_unless (gst_tag_list_get_string (tag_list, GST_TAG_ARTIST, &artist));
- fail_unless (g_str_equal (artist, "Foobar"));
- g_free (artist);
- gst_event_unref (new_event);
-
- /* Same as above, but with a non-writable event. */
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_TRACK_GAIN, +4.95, GST_TAG_TRACK_PEAK, 0.59463,
- GST_TAG_ALBUM_GAIN, -1.54, GST_TAG_ALBUM_PEAK, 0.693415,
- GST_TAG_ARTIST, "Foobar", NULL);
- event = gst_event_new_tag (tag_list);
- /* Holding an extra ref makes the event unwritable: */
- gst_event_ref (event);
- new_event = send_tag_event (element, event);
- fail_unless (event != new_event, "Unwritable tag event reused");
- gst_event_parse_tag (new_event, &tag_list);
- fail_unless (gst_tag_list_get_string (tag_list, GST_TAG_ARTIST, &artist));
- fail_unless (g_str_equal (artist, "Foobar"));
- g_free (artist);
- gst_event_unref (event);
- gst_event_unref (new_event);
-
- cleanup_rgvolume (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_simple)
-{
- GstElement *element = setup_rgvolume ();
- GstTagList *tag_list;
-
- g_object_set (element, "album-mode", FALSE, "headroom", +0.00,
- "pre-amp", -6.00, "fallback-gain", +1.23, NULL);
- set_playing_state (element);
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_TRACK_GAIN, -3.45, GST_TAG_TRACK_PEAK, 1.0,
- GST_TAG_ALBUM_GAIN, +2.09, GST_TAG_ALBUM_PEAK, 1.0, NULL);
- fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
- fail_unless_gain (element, -9.45); /* pre-amp + track gain */
- send_eos_event (element);
-
- g_object_set (element, "album-mode", TRUE, NULL);
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_TRACK_GAIN, -3.45, GST_TAG_TRACK_PEAK, 1.0,
- GST_TAG_ALBUM_GAIN, +2.09, GST_TAG_ALBUM_PEAK, 1.0, NULL);
- fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
- fail_unless_gain (element, -3.91); /* pre-amp + album gain */
-
- /* Switching back to track mode in the middle of a stream: */
- g_object_set (element, "album-mode", FALSE, NULL);
- fail_unless_gain (element, -9.45); /* pre-amp + track gain */
- send_eos_event (element);
-
- cleanup_rgvolume (element);
-}
-
-GST_END_TEST;
-
-/* If there are no gain tags at all, the fallback gain is used. */
-
-GST_START_TEST (test_fallback_gain)
-{
- GstElement *element = setup_rgvolume ();
- GstTagList *tag_list;
-
- /* First some track where fallback does _not_ apply. */
-
- g_object_set (element, "album-mode", FALSE, "headroom", 10.00,
- "pre-amp", -6.00, "fallback-gain", -3.00, NULL);
- set_playing_state (element);
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_TRACK_GAIN, +3.5, GST_TAG_TRACK_PEAK, 1.0,
- GST_TAG_ALBUM_GAIN, -0.5, GST_TAG_ALBUM_PEAK, 1.0, NULL);
- fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
- fail_unless_gain (element, -2.50); /* pre-amp + track gain */
- send_eos_event (element);
-
- /* Now a track completely missing tags. */
-
- fail_unless_gain (element, -9.00); /* pre-amp + fallback-gain */
-
- /* Changing the fallback gain in the middle of a stream, going to pass-through
- * mode: */
- g_object_set (element, "fallback-gain", +6.00, NULL);
- fail_unless_gain (element, +0.00); /* pre-amp + fallback-gain */
- send_eos_event (element);
-
- /* Verify that result gain is set to +0.00 with pre-amp + fallback-gain >
- * +0.00 and no headroom. */
-
- g_object_set (element, "fallback-gain", +12.00, "headroom", +0.00, NULL);
- fail_unless_target_gain (element, +6.00); /* pre-amp + fallback-gain */
- fail_unless_result_gain (element, +0.00);
- send_eos_event (element);
-
- cleanup_rgvolume (element);
-}
-
-GST_END_TEST;
-
-/* If album gain is to be preferred but not available, the track gain is to be
- * taken instead. */
-
-GST_START_TEST (test_fallback_track)
-{
- GstElement *element = setup_rgvolume ();
- GstTagList *tag_list;
-
- g_object_set (element, "album-mode", TRUE, "headroom", +0.00,
- "pre-amp", -6.00, "fallback-gain", +1.23, NULL);
- set_playing_state (element);
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_TRACK_GAIN, +2.11, GST_TAG_TRACK_PEAK, 1.0, NULL);
- fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
- fail_unless_gain (element, -3.89); /* pre-amp + track gain */
-
- send_eos_event (element);
-
- cleanup_rgvolume (element);
-}
-
-GST_END_TEST;
-
-/* If track gain is to be preferred but not available, the album gain is to be
- * taken instead. */
-
-GST_START_TEST (test_fallback_album)
-{
- GstElement *element = setup_rgvolume ();
- GstTagList *tag_list;
-
- g_object_set (element, "album-mode", FALSE, "headroom", +0.00,
- "pre-amp", -6.00, "fallback-gain", +1.23, NULL);
- set_playing_state (element);
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_ALBUM_GAIN, +3.73, GST_TAG_ALBUM_PEAK, 1.0, NULL);
- fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
- fail_unless_gain (element, -2.27); /* pre-amp + album gain */
-
- send_eos_event (element);
-
- cleanup_rgvolume (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_headroom)
-{
- GstElement *element = setup_rgvolume ();
- GstTagList *tag_list;
-
- g_object_set (element, "album-mode", FALSE, "headroom", +0.00,
- "pre-amp", +0.00, "fallback-gain", +1.23, NULL);
- set_playing_state (element);
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_TRACK_GAIN, +3.50, GST_TAG_TRACK_PEAK, 1.0, NULL);
- fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
- fail_unless_target_gain (element, +3.50); /* pre-amp + track gain */
- fail_unless_result_gain (element, +0.00);
- send_eos_event (element);
-
- g_object_set (element, "headroom", +2.00, NULL);
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_TRACK_GAIN, +9.18, GST_TAG_TRACK_PEAK, 0.687149, NULL);
- fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
- fail_unless_target_gain (element, +9.18); /* pre-amp + track gain */
- /* Result is 20. * log10 (1. / peak) + headroom. */
- fail_unless_result_gain (element, 5.2589816238303335);
- send_eos_event (element);
-
- g_object_set (element, "album-mode", TRUE, NULL);
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_ALBUM_GAIN, +5.50, GST_TAG_ALBUM_PEAK, 1.0, NULL);
- fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
- fail_unless_target_gain (element, +5.50); /* pre-amp + album gain */
- fail_unless_result_gain (element, +2.00); /* headroom */
- send_eos_event (element);
-
- cleanup_rgvolume (element);
-}
-
-GST_END_TEST;
-
-GST_START_TEST (test_reference_level)
-{
- GstElement *element = setup_rgvolume ();
- GstTagList *tag_list;
-
- g_object_set (element,
- "album-mode", FALSE,
- "headroom", +0.00, "pre-amp", +0.00, "fallback-gain", +1.23, NULL);
- set_playing_state (element);
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_TRACK_GAIN, 0.00, GST_TAG_TRACK_PEAK, 0.2,
- GST_TAG_REFERENCE_LEVEL, 83., NULL);
- fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
- /* Because our authorative reference is 89 dB, we bump it up by +6 dB. */
- fail_unless_gain (element, +6.00); /* pre-amp + track gain */
- send_eos_event (element);
-
- g_object_set (element, "album-mode", TRUE, NULL);
-
- /* Same as above, but with album gain. */
-
- tag_list = gst_tag_list_new ();
- gst_tag_list_add (tag_list, GST_TAG_MERGE_REPLACE,
- GST_TAG_TRACK_GAIN, 1.23, GST_TAG_TRACK_PEAK, 0.1,
- GST_TAG_ALBUM_GAIN, 0.00, GST_TAG_ALBUM_PEAK, 0.2,
- GST_TAG_REFERENCE_LEVEL, 83., NULL);
- fail_unless (send_tag_event (element, gst_event_new_tag (tag_list)) == NULL);
- fail_unless_gain (element, +6.00); /* pre-amp + album gain */
-
- cleanup_rgvolume (element);
-}
-
-GST_END_TEST;
-
-Suite *
-rgvolume_suite (void)
-{
- Suite *s = suite_create ("rgvolume");
- TCase *tc_chain = tcase_create ("general");
-
- suite_add_tcase (s, tc_chain);
-
- tcase_add_test (tc_chain, test_no_buffer);
- tcase_add_test (tc_chain, test_events);
- tcase_add_test (tc_chain, test_simple);
- tcase_add_test (tc_chain, test_fallback_gain);
- tcase_add_test (tc_chain, test_fallback_track);
- tcase_add_test (tc_chain, test_fallback_album);
- tcase_add_test (tc_chain, test_headroom);
- tcase_add_test (tc_chain, test_reference_level);
-
- return s;
-}
-
-int
-main (int argc, char **argv)
-{
- gint nf;
-
- Suite *s = rgvolume_suite ();
- SRunner *sr = srunner_create (s);
-
- gst_check_init (&argc, &argv);
-
- srunner_run_all (sr, CK_ENV);
- nf = srunner_ntests_failed (sr);
- srunner_free (sr);
-
- return nf;
-}
projects / platform / upstream / gst-plugins-bad.git / commitdiff