diff options
| author | Jan Schmidt <thaytan@mad.scientist.com> | 2008-07-19 00:58:49 +0000 |
|---|---|---|
| committer | Jan Schmidt <thaytan@mad.scientist.com> | 2008-07-19 00:58:49 +0000 |
| commit | e985585a4ec8ec1a681c9643f6727a230fb536d7 (patch) | |
| tree | 1b1fc2eeabad64ca5ba42b51cf42acc082686189 | |
| parent | 26cb95316c8043e05365337660c1e07b067f298e (diff) | |
| download | gst-plugins-bad-e985585a4ec8ec1a681c9643f6727a230fb536d7.tar.gz gst-plugins-bad-e985585a4ec8ec1a681c9643f6727a230fb536d7.tar.bz2 gst-plugins-bad-e985585a4ec8ec1a681c9643f6727a230fb536d7.zip | |
Remove interleave and replaygain plugins that have moved to -good
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.hierarchy:
* docs/plugins/gst-plugins-bad-plugins.interfaces:
* docs/plugins/gst-plugins-bad-plugins.prerequisites:
* docs/plugins/inspect/plugin-interleave.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* gst/interleave/Makefile.am:
* gst/interleave/deinterleave.c:
* gst/interleave/deinterleave.h:
* gst/interleave/interleave.c:
* gst/interleave/interleave.h:
* gst/interleave/plugin.c:
* gst/interleave/plugin.h:
* gst/replaygain/Makefile.am:
* gst/replaygain/gstrganalysis.c:
* gst/replaygain/gstrganalysis.h:
* gst/replaygain/gstrglimiter.c:
* gst/replaygain/gstrglimiter.h:
* gst/replaygain/gstrgvolume.c:
* gst/replaygain/gstrgvolume.h:
* gst/replaygain/replaygain.c:
* gst/replaygain/replaygain.h:
* gst/replaygain/rganalysis.c:
* gst/replaygain/rganalysis.h:
* tests/check/Makefile.am:
* tests/check/elements/deinterleave.c:
* tests/check/elements/interleave.c:
* tests/check/elements/rganalysis.c:
* tests/check/elements/rglimiter.c:
* tests/check/elements/rgvolume.c:
Remove interleave and replaygain plugins that have moved to -good
31 files changed, 40 insertions, 9573 deletions
@@ -1,3 +1,40 @@ +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: diff --git a/docs/plugins/Makefile.am b/docs/plugins/Makefile.am index d3a34043..ba2361f5 100644 --- a/docs/plugins/Makefile.am +++ b/docs/plugins/Makefile.am @@ -115,15 +115,10 @@ EXTRA_HFILES = \ $(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 \ diff --git a/docs/plugins/gst-plugins-bad-plugins-docs.sgml b/docs/plugins/gst-plugins-bad-plugins-docs.sgml index ce5ad0f5..ba5d3b1a 100644 --- a/docs/plugins/gst-plugins-bad-plugins-docs.sgml +++ b/docs/plugins/gst-plugins-bad-plugins-docs.sgml @@ -17,7 +17,6 @@ <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" /> @@ -29,7 +28,6 @@ <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" /> @@ -39,9 +37,6 @@ <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" /> @@ -84,7 +79,6 @@ <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" /> @@ -103,7 +97,6 @@ <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" /> diff --git a/docs/plugins/gst-plugins-bad-plugins-sections.txt b/docs/plugins/gst-plugins-bad-plugins-sections.txt index 54af85a9..57a97640 100644 --- a/docs/plugins/gst-plugins-bad-plugins-sections.txt +++ b/docs/plugins/gst-plugins-bad-plugins-sections.txt @@ -168,6 +168,7 @@ FESTIVAL_DEFAULT_SERVER_PORT FESTIVAL_DEFAULT_TEXT_MODE </SECTION> +<SECTION> <FILE>element-input-selector</FILE> <TITLE>input-selector</TITLE> GstInputSelector @@ -202,38 +203,6 @@ 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 <TITLE>jackaudiosink</TITLE> @@ -381,48 +350,6 @@ 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> GstRtpBin diff --git a/docs/plugins/gst-plugins-bad-plugins.prerequisites b/docs/plugins/gst-plugins-bad-plugins.prerequisites index b703e314..bb5fd3fc 100644 --- a/docs/plugins/gst-plugins-bad-plugins.prerequisites +++ b/docs/plugins/gst-plugins-bad-plugins.prerequisites @@ -2,4 +2,6 @@ GstChildProxy GstObject GstTagSetter GstObject GstElement GstImplementsInterface GstObject GstElement GstXOverlay GstObject GstImplementsInterface GstElement +GstTagSetter GstObject GstElement +GstColorBalance GstObject GstImplementsInterface GstElement GstMixer GstObject GstImplementsInterface GstElement diff --git a/docs/plugins/inspect/plugin-interleave.xml b/docs/plugins/inspect/plugin-interleave.xml deleted file mode 100644 index b31ad555..00000000 --- a/docs/plugins/inspect/plugin-interleave.xml +++ /dev/null @@ -1,55 +0,0 @@ -<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 diff --git a/docs/plugins/inspect/plugin-replaygain.xml b/docs/plugins/inspect/plugin-replaygain.xml deleted file mode 100644 index b45ffc1d..00000000 --- a/docs/plugins/inspect/plugin-replaygain.xml +++ /dev/null @@ -1,76 +0,0 @@ -<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 diff --git a/gst/interleave/Makefile.am b/gst/interleave/Makefile.am deleted file mode 100644 index 3477933c..00000000 --- a/gst/interleave/Makefile.am +++ /dev/null @@ -1,9 +0,0 @@ - -plugin_LTLIBRARIES = libgstinterleave.la - -libgstinterleave_la_SOURCES = plugin.c interleave.c deinterleave.c -libgstinterleave_la_CFLAGS = $(GST_CFLAGS) $(GST_BASE_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) -libgstinterleave_la_LIBADD = $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstaudio-$(GST_MAJORMINOR) -libgstinterleave_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) - -noinst_HEADERS = plugin.h interleave.h deinterleave.h diff --git a/gst/interleave/deinterleave.c b/gst/interleave/deinterleave.c deleted file mode 100644 index 4c81d39d..00000000 --- a/gst/interleave/deinterleave.c +++ /dev/null @@ -1,889 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2000 Wim Taymans <wtay@chello.be> - * 2005 Wim Taymans <wim@fluendo.com> - * 2007 Andy Wingo <wingo at pobox.com> - * 2008 Sebastian Dröge <slomo@circular-chaos.org> - * - * deinterleave.c: deinterleave samples - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/* TODO: - * - handle changes in number of channels - * - handle changes in channel positions - * - better capsnego by using a buffer alloc function - * and passing downstream caps changes upstream there - */ - -/** - * SECTION:element-deinterleave - * @see_also: interleave - * - * Splits one interleaved multichannel audio stream into many mono audio streams. - * - * This element handles all raw audio formats and supports changing the input caps as long as - * all downstream elements can handle the new caps and the number of channels and the channel - * positions stay the same. This restriction will be removed in later versions by adding or - * removing some source pads as required. - * - * In most cases a queue and an audioconvert element should be added after each source pad - * before further processing of the audio data. - * - * <refsect2> - * <title>Example launch line</title> - * |[ - * gst-launch-0.10 filesrc location=/path/to/file.mp3 ! decodebin ! audioconvert ! "audio/x-raw-int,channels=2 ! deinterleave name=d d.src0 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel1.ogg d.src1 ! queue ! audioconvert ! vorbisenc ! oggmux ! filesink location=channel2.ogg - * ]| Decodes an MP3 file and encodes the left and right channel into separate - * Ogg Vorbis files. - * |[ - * gst-launch-0.10 filesrc location=file.mp3 ! decodebin ! audioconvert ! "audio/x-raw-int,channels=2" ! deinterleave name=d interleave name=i ! audioconvert ! wavenc ! filesink location=test.wav d.src0 ! queue ! audioconvert ! i.sink1 d.src1 ! queue ! audioconvert ! i.sink0 - * ]| Decodes and deinterleaves a Stereo MP3 file into separate channels and - * then interleaves the channels again to a WAV file with the channel with the - * channels exchanged. - * </refsect2> - */ - -#ifdef HAVE_CONFIG_H -# include "config.h" -#endif - -#include <gst/gst.h> -#include <string.h> -#include "deinterleave.h" - -GST_DEBUG_CATEGORY_STATIC (gst_deinterleave_debug); -#define GST_CAT_DEFAULT gst_deinterleave_debug - -static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src%d", - GST_PAD_SRC, - GST_PAD_SOMETIMES, - GST_STATIC_CAPS ("audio/x-raw-int, " - "rate = (int) [ 1, MAX ], " - "channels = (int) 1, " - "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " - "width = (int) { 8, 16, 24, 32 }, " - "depth = (int) [ 1, 32 ], " - "signed = (boolean) { true, false }; " - "audio/x-raw-float, " - "rate = (int) [ 1, MAX ], " - "channels = (int) 1, " - "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, " - "width = (int) { 32, 64 }") - ); - -static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-int, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, MAX ], " - "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " - "width = (int) { 8, 16, 24, 32 }, " - "depth = (int) [ 1, 32 ], " - "signed = (boolean) { true, false }; " - "audio/x-raw-float, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, MAX ], " - "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, " - "width = (int) { 32, 64 }") - ); - -#define MAKE_FUNC(type) \ -static void deinterleave_##type (guint##type *out, guint##type *in, \ - guint stride, guint nframes) \ -{ \ - gint i; \ - \ - for (i = 0; i < nframes; i++) { \ - out[i] = *in; \ - in += stride; \ - } \ -} - -MAKE_FUNC (8); -MAKE_FUNC (16); -MAKE_FUNC (32); -MAKE_FUNC (64); - -static void -deinterleave_24 (guint8 * out, guint8 * in, guint stride, guint nframes) -{ - gint i; - - for (i = 0; i < nframes; i++) { - memcpy (out, in, 3); - out += 3; - in += stride * 3; - } -} - -GST_BOILERPLATE (GstDeinterleave, gst_deinterleave, GstElement, - GST_TYPE_ELEMENT); - -enum -{ - PROP_0, - PROP_KEEP_POSITIONS -}; - -static GstFlowReturn gst_deinterleave_chain (GstPad * pad, GstBuffer * buffer); - -static gboolean gst_deinterleave_sink_setcaps (GstPad * pad, GstCaps * caps); - -static GstCaps *gst_deinterleave_sink_getcaps (GstPad * pad); - -static gboolean gst_deinterleave_sink_activate_push (GstPad * pad, - gboolean active); -static gboolean gst_deinterleave_sink_event (GstPad * pad, GstEvent * event); - -static gboolean gst_deinterleave_src_query (GstPad * pad, GstQuery * query); - -static void gst_deinterleave_set_property (GObject * object, - guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_deinterleave_get_property (GObject * object, - guint prop_id, GValue * value, GParamSpec * pspec); - - -static void -gst_deinterleave_finalize (GObject * obj) -{ - GstDeinterleave *self = GST_DEINTERLEAVE (obj); - - if (self->pos) { - g_free (self->pos); - self->pos = NULL; - } - - if (self->pending_events) { - g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref, NULL); - g_list_free (self->pending_events); - self->pending_events = NULL; - } - - G_OBJECT_CLASS (parent_class)->finalize (obj); -} - -static void -gst_deinterleave_base_init (gpointer g_class) -{ - GstElementClass *gstelement_class = (GstElementClass *) g_class; - - gst_element_class_set_details_simple (gstelement_class, "Audio deinterleaver", - "Filter/Converter/Audio", - "Splits one interleaved multichannel audio stream into many mono audio streams", - "Andy Wingo <wingo at pobox.com>, " - "Iain <iain@prettypeople.org>, " - "Sebastian Dröge <slomo@circular-chaos.org>"); - - gst_element_class_add_pad_template (gstelement_class, - gst_static_pad_template_get (&sink_template)); - gst_element_class_add_pad_template (gstelement_class, - gst_static_pad_template_get (&src_template)); -} - -static void -gst_deinterleave_class_init (GstDeinterleaveClass * klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - - GST_DEBUG_CATEGORY_INIT (gst_deinterleave_debug, "deinterleave", 0, - "deinterleave element"); - - gobject_class->finalize = gst_deinterleave_finalize; - gobject_class->set_property = gst_deinterleave_set_property; - gobject_class->get_property = gst_deinterleave_get_property; - - /** - * GstDeinterleave:keep-positions - * - * Keep positions: When enable the caps on the output buffers will - * contain the original channel positions. This can be used to correctly - * interleave the output again later but can also lead to unwanted effects - * if the output should be handled as Mono. - * - */ - g_object_class_install_property (gobject_class, PROP_KEEP_POSITIONS, - g_param_spec_boolean ("keep-positions", "Keep positions", - "Keep the original channel positions on the output buffers", - FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); -} - -static void -gst_deinterleave_init (GstDeinterleave * self, GstDeinterleaveClass * klass) -{ - self->channels = 0; - self->pos = NULL; - self->keep_positions = FALSE; - self->width = 0; - self->func = NULL; - - /* Add sink pad */ - self->sink = gst_pad_new_from_static_template (&sink_template, "sink"); - gst_pad_set_chain_function (self->sink, - GST_DEBUG_FUNCPTR (gst_deinterleave_chain)); - gst_pad_set_setcaps_function (self->sink, - GST_DEBUG_FUNCPTR (gst_deinterleave_sink_setcaps)); - gst_pad_set_getcaps_function (self->sink, - GST_DEBUG_FUNCPTR (gst_deinterleave_sink_getcaps)); - gst_pad_set_activatepush_function (self->sink, - GST_DEBUG_FUNCPTR (gst_deinterleave_sink_activate_push)); - gst_pad_set_event_function (self->sink, - GST_DEBUG_FUNCPTR (gst_deinterleave_sink_event)); - gst_element_add_pad (GST_ELEMENT (self), self->sink); -} - -static void -gst_deinterleave_add_new_pads (GstDeinterleave * self, GstCaps * caps) -{ - GstPad *pad; - - guint i; - - for (i = 0; i < self->channels; i++) { - gchar *name = g_strdup_printf ("src%d", i); - - GstCaps *srccaps; - - GstStructure *s; - - pad = gst_pad_new_from_static_template (&src_template, name); - g_free (name); - - /* Set channel position if we know it */ - if (self->keep_positions) { - GstAudioChannelPosition pos[1] = { GST_AUDIO_CHANNEL_POSITION_NONE }; - - srccaps = gst_caps_copy (caps); - s = gst_caps_get_structure (srccaps, 0); - if (self->pos) - gst_audio_set_channel_positions (s, &self->pos[i]); - else - gst_audio_set_channel_positions (s, pos); - } else { - srccaps = caps; - } - - gst_pad_set_caps (pad, srccaps); - gst_pad_use_fixed_caps (pad); - gst_pad_set_query_function (pad, - GST_DEBUG_FUNCPTR (gst_deinterleave_src_query)); - gst_pad_set_active (pad, TRUE); - gst_element_add_pad (GST_ELEMENT (self), pad); - self->srcpads = g_list_prepend (self->srcpads, gst_object_ref (pad)); - - if (self->keep_positions) - gst_caps_unref (srccaps); - } - - gst_element_no_more_pads (GST_ELEMENT (self)); - self->srcpads = g_list_reverse (self->srcpads); -} - -static void -gst_deinterleave_set_pads_caps (GstDeinterleave * self, GstCaps * caps) -{ - GList *l; - - GstStructure *s; - - gint i; - - for (l = self->srcpads, i = 0; l; l = l->next, i++) { - GstPad *pad = GST_PAD (l->data); - - GstCaps *srccaps; - - /* Set channel position if we know it */ - if (self->keep_positions) { - GstAudioChannelPosition pos[1] = { GST_AUDIO_CHANNEL_POSITION_NONE }; - - srccaps = gst_caps_copy (caps); - s = gst_caps_get_structure (srccaps, 0); - if (self->pos) - gst_audio_set_channel_positions (s, &self->pos[i]); - else - gst_audio_set_channel_positions (s, pos); - } else { - srccaps = caps; - } - - gst_pad_set_caps (pad, srccaps); - - if (self->keep_positions) - gst_caps_unref (srccaps); - } -} - -static void -gst_deinterleave_remove_pads (GstDeinterleave * self) -{ - GList *l; - - GST_INFO_OBJECT (self, "removing pads"); - - for (l = self->srcpads; l; l = l->next) { - GstPad *pad = GST_PAD (l->data); - - gst_element_remove_pad (GST_ELEMENT_CAST (self), pad); - gst_object_unref (pad); - } - g_list_free (self->srcpads); - self->srcpads = NULL; - - gst_pad_set_caps (self->sink, NULL); - gst_caps_replace (&self->sinkcaps, NULL); -} - -static gboolean -gst_deinterleave_set_process_function (GstDeinterleave * self, GstCaps * caps) -{ - GstStructure *s; - - s = gst_caps_get_structure (caps, 0); - if (!gst_structure_get_int (s, "width", &self->width)) - return FALSE; - - switch (self->width) { - case 8: - self->func = (GstDeinterleaveFunc) deinterleave_8; - break; - case 16: - self->func = (GstDeinterleaveFunc) deinterleave_16; - break; - case 24: - self->func = (GstDeinterleaveFunc) deinterleave_24; - break; - case 32: - self->func = (GstDeinterleaveFunc) deinterleave_32; - break; - case 64: - self->func = (GstDeinterleaveFunc) deinterleave_64; - break; - default: - return FALSE; - } - return TRUE; -} - -static gboolean -gst_deinterleave_sink_setcaps (GstPad * pad, GstCaps * caps) -{ - GstDeinterleave *self; - - GstCaps *srccaps; - - GstStructure *s; - - self = GST_DEINTERLEAVE (gst_pad_get_parent (pad)); - - GST_DEBUG_OBJECT (self, "got caps: %" GST_PTR_FORMAT, caps); - - if (self->sinkcaps && !gst_caps_is_equal (caps, self->sinkcaps)) { - gint new_channels, i; - - GstAudioChannelPosition *pos; - - gboolean same_layout = TRUE; - - s = gst_caps_get_structure (caps, 0); - - /* We allow caps changes as long as the number of channels doesn't change - * and the channel positions stay the same. _getcaps() should've cared - * for this already but better be safe. - */ - if (!gst_structure_get_int (s, "channels", &new_channels) || - new_channels != self->channels || - !gst_deinterleave_set_process_function (self, caps)) - goto cannot_change_caps; - - /* Now check the channel positions. If we had no channel positions - * and get them or the other way around things have changed. - * If we had channel positions and get different ones things have - * changed too of course - */ - pos = gst_audio_get_channel_positions (s); - if ((pos && !self->pos) || (!pos && self->pos)) - goto cannot_change_caps; - - if (pos) { - for (i = 0; i < self->channels; i++) { - if (self->pos[i] != pos[i]) { - same_layout = FALSE; - break; - } - } - g_free (pos); - if (!same_layout) - goto cannot_change_caps; - } - - } else { - s = gst_caps_get_structure (caps, 0); - - if (!gst_structure_get_int (s, "channels", &self->channels)) - goto no_channels; - - if (!gst_deinterleave_set_process_function (self, caps)) - goto unsupported_caps; - - self->pos = gst_audio_get_channel_positions (s); - } - - gst_caps_replace (&self->sinkcaps, caps); - - /* Get srcpad caps */ - srccaps = gst_caps_copy (caps); - s = gst_caps_get_structure (srccaps, 0); - gst_structure_set (s, "channels", G_TYPE_INT, 1, NULL); - gst_structure_remove_field (s, "channel-positions"); - - /* If we already have pads, update the caps otherwise - * add new pads */ - if (self->srcpads) { - gst_deinterleave_set_pads_caps (self, srccaps); - } else { - gst_deinterleave_add_new_pads (self, srccaps); - } - - gst_caps_unref (srccaps); - gst_object_unref (self); - - return TRUE; - -cannot_change_caps: - { - GST_ERROR_OBJECT (self, "can't set new caps: %" GST_PTR_FORMAT, caps); - gst_object_unref (self); - return FALSE; - } -unsupported_caps: - { - GST_ERROR_OBJECT (self, "caps not supported: %" GST_PTR_FORMAT, caps); - gst_object_unref (self); - return FALSE; - } -no_channels: - { - GST_ERROR_OBJECT (self, "invalid caps"); - gst_object_unref (self); - return FALSE; - } -} - -static void -__remove_channels (GstCaps * caps) -{ - GstStructure *s; - - gint i, size; - - size = gst_caps_get_size (caps); - for (i = 0; i < size; i++) { - s = gst_caps_get_structure (caps, i); - gst_structure_remove_field (s, "channel-positions"); - gst_structure_remove_field (s, "channels"); - } -} - -static void -__set_channels (GstCaps * caps, gint channels) -{ - GstStructure *s; - - gint i, size; - - size = gst_caps_get_size (caps); - for (i = 0; i < size; i++) { - s = gst_caps_get_structure (caps, i); - if (channels > 0) - gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL); - else - gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); - } -} - -static GstCaps * -gst_deinterleave_sink_getcaps (GstPad * pad) -{ - GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad)); - - GstCaps *ret; - - GList *l; - - GST_OBJECT_LOCK (self); - /* Intersect all of our pad template caps with the peer caps of the pad - * to get all formats that are possible up- and downstream. - * - * For the pad for which the caps are requested we don't remove the channel - * informations as they must be in the returned caps and incompatibilities - * will be detected here already - */ - ret = gst_caps_new_any (); - for (l = GST_ELEMENT (self)->pads; l != NULL; l = l->next) { - GstPad *ourpad = GST_PAD (l->data); - - GstCaps *peercaps = NULL, *ourcaps; - - ourcaps = gst_caps_copy (gst_pad_get_pad_template_caps (ourpad)); - - if (pad == ourpad) { - if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) - __set_channels (ourcaps, self->channels); - else - __set_channels (ourcaps, 1); - } else { - __remove_channels (ourcaps); - /* Only ask for peer caps for other pads than pad - * as otherwise gst_pad_peer_get_caps() might call - * back into this function and deadlock - */ - peercaps = gst_pad_peer_get_caps (ourpad); - } - - /* If the peer exists and has caps add them to the intersection, - * otherwise assume that the peer accepts everything */ - if (peercaps) { - GstCaps *intersection; - - GstCaps *oldret = ret; - - __remove_channels (peercaps); - - intersection = gst_caps_intersect (peercaps, ourcaps); - - ret = gst_caps_intersect (ret, intersection); - gst_caps_unref (intersection); - gst_caps_unref (peercaps); - gst_caps_unref (oldret); - } else { - GstCaps *oldret = ret; - - ret = gst_caps_intersect (ret, ourcaps); - gst_caps_unref (oldret); - } - gst_caps_unref (ourcaps); - } - GST_OBJECT_UNLOCK (self); - - gst_object_unref (self); - - GST_DEBUG_OBJECT (pad, "Intersected caps to %" GST_PTR_FORMAT, ret); - - return ret; -} - -static gboolean -gst_deinterleave_sink_event (GstPad * pad, GstEvent * event) -{ - GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad)); - - gboolean ret; - - GST_DEBUG ("Got %s event on pad %s:%s", GST_EVENT_TYPE_NAME (event), - GST_DEBUG_PAD_NAME (pad)); - - /* Send FLUSH_STOP, FLUSH_START and EOS immediately, no matter if - * we have src pads already or not. Queue all other events and - * push them after we have src pads - */ - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_FLUSH_STOP: - case GST_EVENT_FLUSH_START: - case GST_EVENT_EOS: - ret = gst_pad_event_default (pad, event); - break; - default: - if (self->srcpads) { - ret = gst_pad_event_default (pad, event); - } else { - GST_OBJECT_LOCK (self); - self->pending_events = g_list_append (self->pending_events, event); - GST_OBJECT_UNLOCK (self); - ret = TRUE; - } - break; - } - - gst_object_unref (self); - - return ret; -} - -static gboolean -gst_deinterleave_src_query (GstPad * pad, GstQuery * query) -{ - GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad)); - - gboolean res; - - res = gst_pad_query_default (pad, query); - - if (res && GST_QUERY_TYPE (query) == GST_QUERY_DURATION) { - GstFormat format; - - gint64 dur; - - gst_query_parse_duration (query, &format, &dur); - - /* Need to divide by the number of channels in byte format - * to get the correct value. All other formats should be fine - */ - if (format == GST_FORMAT_BYTES && dur != -1) - gst_query_set_duration (query, format, dur / self->channels); - } else if (res && GST_QUERY_TYPE (query) == GST_QUERY_POSITION) { - GstFormat format; - - gint64 pos; - - gst_query_parse_position (query, &format, &pos); - - /* Need to divide by the number of channels in byte format - * to get the correct value. All other formats should be fine - */ - if (format == GST_FORMAT_BYTES && pos != -1) - gst_query_set_position (query, format, pos / self->channels); - } - - gst_object_unref (self); - return res; -} - -static void -gst_deinterleave_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstDeinterleave *self = GST_DEINTERLEAVE (object); - - switch (prop_id) { - case PROP_KEEP_POSITIONS: - self->keep_positions = g_value_get_boolean (value); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_deinterleave_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstDeinterleave *self = GST_DEINTERLEAVE (object); - - switch (prop_id) { - case PROP_KEEP_POSITIONS: - g_value_set_boolean (value, self->keep_positions); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static GstFlowReturn -gst_deinterleave_process (GstDeinterleave * self, GstBuffer * buf) -{ - GstFlowReturn ret = GST_FLOW_OK; - - guint channels = self->channels; - - guint pads_pushed = 0, buffers_allocated = 0; - - guint nframes = GST_BUFFER_SIZE (buf) / channels / (self->width / 8); - - guint bufsize = nframes * (self->width / 8); - - guint i; - - GList *srcs; - - GstBuffer **buffers_out = g_new0 (GstBuffer *, channels); - - guint8 *in, *out; - - /* Send any pending events to all src pads */ - GST_OBJECT_LOCK (self); - if (self->pending_events) { - GList *events; - - GstEvent *event; - - GST_DEBUG_OBJECT (self, "Sending pending events to all src pads"); - - for (events = self->pending_events; events != NULL; events = events->next) { - event = GST_EVENT (events->data); - - for (srcs = self->srcpads; srcs != NULL; srcs = srcs->next) - gst_pad_push_event (GST_PAD (srcs->data), gst_event_ref (event)); - gst_event_unref (event); - } - - g_list_free (self->pending_events); - self->pending_events = NULL; - } - GST_OBJECT_UNLOCK (self); - - /* Allocate buffers */ - for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) { - GstPad *pad = (GstPad *) srcs->data; - - buffers_out[i] = NULL; - ret = - gst_pad_alloc_buffer (pad, GST_BUFFER_OFFSET_NONE, bufsize, - GST_PAD_CAPS (pad), &buffers_out[i]); - - /* Make sure we got a correct buffer. The only other case we allow - * here is an unliked pad */ - if (ret != GST_FLOW_OK && ret != GST_FLOW_NOT_LINKED) - goto alloc_buffer_failed; - else if (buffers_out[i] && GST_BUFFER_SIZE (buffers_out[i]) != bufsize) - goto alloc_buffer_bad_size; - else if (buffers_out[i] && - !gst_caps_is_equal (GST_BUFFER_CAPS (buffers_out[i]), - GST_PAD_CAPS (pad))) - goto invalid_caps; - - if (buffers_out[i]) { - gst_buffer_copy_metadata (buffers_out[i], buf, - GST_BUFFER_COPY_TIMESTAMPS | GST_BUFFER_COPY_FLAGS); - buffers_allocated++; - } - } - - /* Return NOT_LINKED if no pad was linked */ - if (!buffers_allocated) { - GST_WARNING_OBJECT (self, - "Couldn't allocate any buffers because no pad was linked"); - ret = GST_FLOW_NOT_LINKED; - goto done; - } - - /* deinterleave */ - for (srcs = self->srcpads, i = 0; srcs; srcs = srcs->next, i++) { - GstPad *pad = (GstPad *) srcs->data; - - in = (guint8 *) GST_BUFFER_DATA (buf); - in += i * (self->width / 8); - if (buffers_out[i]) { - out = (guint8 *) GST_BUFFER_DATA (buffers_out[i]); - - self->func (out, in, channels, nframes); - - ret = gst_pad_push (pad, buffers_out[i]); - buffers_out[i] = NULL; - if (ret == GST_FLOW_OK) - pads_pushed++; - else if (ret == GST_FLOW_NOT_LINKED) - ret = GST_FLOW_OK; - else - goto push_failed; - } - } - - /* Return NOT_LINKED if no pad was linked */ - if (!pads_pushed) - ret = GST_FLOW_NOT_LINKED; - -done: - gst_buffer_unref (buf); - g_free (buffers_out); - return ret; - -alloc_buffer_failed: - { - GST_WARNING ("gst_pad_alloc_buffer() returned %s", gst_flow_get_name (ret)); - goto clean_buffers; - - } -alloc_buffer_bad_size: - { - GST_WARNING ("called alloc_buffer(), but didn't get requested bytes"); - ret = GST_FLOW_NOT_NEGOTIATED; - goto clean_buffers; - } -invalid_caps: - { - GST_WARNING ("called alloc_buffer(), but didn't get requested caps"); - ret = GST_FLOW_NOT_NEGOTIATED; - goto clean_buffers; - } -push_failed: - { - GST_DEBUG ("push() failed, flow = %s", gst_flow_get_name (ret)); - goto clean_buffers; - } -clean_buffers: - { - for (i = 0; i < channels; i++) { - if (buffers_out[i]) - gst_buffer_unref (buffers_out[i]); - } - gst_buffer_unref (buf); - g_free (buffers_out); - return ret; - } -} - -static GstFlowReturn -gst_deinterleave_chain (GstPad * pad, GstBuffer * buffer) -{ - GstDeinterleave *self = GST_DEINTERLEAVE (GST_PAD_PARENT (pad)); - - GstFlowReturn ret; - - g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED); - g_return_val_if_fail (self->width > 0, GST_FLOW_NOT_NEGOTIATED); - g_return_val_if_fail (self->channels > 0, GST_FLOW_NOT_NEGOTIATED); - - ret = gst_deinterleave_process (self, buffer); - - if (ret != GST_FLOW_OK) - GST_DEBUG_OBJECT (self, "flow return: %s", gst_flow_get_name (ret)); - - return ret; -} - -static gboolean -gst_deinterleave_sink_activate_push (GstPad * pad, gboolean active) -{ - GstDeinterleave *self = GST_DEINTERLEAVE (gst_pad_get_parent (pad)); - - /* Reset everything when the pad is deactivated */ - if (!active) { - gst_deinterleave_remove_pads (self); - if (self->pos) { - g_free (self->pos); - self->pos = NULL; - } - self->channels = 0; - self->width = 0; - self->func = NULL; - - if (self->pending_events) { - g_list_foreach (self->pending_events, (GFunc) gst_mini_object_unref, - NULL); - g_list_free (self->pending_events); - self->pending_events = NULL; - } - } - - gst_object_unref (self); - - return TRUE; -} diff --git a/gst/interleave/deinterleave.h b/gst/interleave/deinterleave.h deleted file mode 100644 index fe8ec75d..00000000 --- a/gst/interleave/deinterleave.h +++ /dev/null @@ -1,75 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2000 Wim Taymans <wtay@chello.be> - * 2005 Wim Taymans <wim@fluendo.com> - * 2007 Andy Wingo <wingo at pobox.com> - * 2008 Sebastian Dröge <slomo@circular-chaos.org> - * - * deinterleave.c: deinterleave samples - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __DEINTERLEAVE_H__ -#define __DEINTERLEAVE_H__ - -G_BEGIN_DECLS - -#include <gst/gst.h> -#include <gst/audio/multichannel.h> - -#define GST_TYPE_DEINTERLEAVE (gst_deinterleave_get_type()) -#define GST_DEINTERLEAVE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_DEINTERLEAVE,GstDeinterleave)) -#define GST_DEINTERLEAVE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_DEINTERLEAVE,GstDeinterleaveClass)) -#define GST_DEINTERLEAVE_GET_CLASS(obj) \ - (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_DEINTERLEAVE,GstDeinterleaveClass)) -#define GST_IS_DEINTERLEAVE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_DEINTERLEAVE)) -#define GST_IS_DEINTERLEAVE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_DEINTERLEAVE)) - -typedef struct _GstDeinterleave GstDeinterleave; -typedef struct _GstDeinterleaveClass GstDeinterleaveClass; - -typedef void (*GstDeinterleaveFunc) (gpointer out, gpointer in, guint stride, guint nframes); - -struct _GstDeinterleave -{ - GstElement element; - - /*< private > */ - GList *srcpads; - GstCaps *sinkcaps; - gint channels; - GstAudioChannelPosition *pos; - gboolean keep_positions; - - GstPad *sink; - - gint width; - GstDeinterleaveFunc func; - - GList *pending_events; -}; - -struct _GstDeinterleaveClass -{ - GstElementClass parent_class; -}; - -GType gst_deinterleave_get_type (void); - -G_END_DECLS - -#endif /* __DEINTERLEAVE_H__ */ diff --git a/gst/interleave/interleave.c b/gst/interleave/interleave.c deleted file mode 100644 index 831e928f..00000000 --- a/gst/interleave/interleave.c +++ /dev/null @@ -1,1352 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2000 Wim Taymans <wtay@chello.be> - * 2005 Wim Taymans <wim@fluendo.com> - * 2007 Andy Wingo <wingo at pobox.com> - * 2008 Sebastian Dröge <slomo@circular-chaos.rg> - * - * interleave.c: interleave samples, mostly based on adder. - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -/* TODO: - * - handle caps changes - * - handle more queries/events - */ - -/** - * SECTION:element-interleave - * @see_also: deinterleave - * - * Merges separate mono inputs into one interleaved stream. - * - * This element handles all raw floating point sample formats and all signed integer sample formats. The first - * caps on one of the sinkpads will set the caps of the output so usually an audioconvert element should be - * placed before every sinkpad of interleave. - * - * It's possible to change the number of channels while the pipeline is running by adding or removing - * some of the request pads but this will change the caps of the output buffers. Changing the input - * caps is _not_ supported yet. - * - * The channel number of every sinkpad in the out can be retrieved from the "channel" property of the pad. - * - * <refsect2> - * <title>Example launch line</title> - * |[ - * gst-launch-0.10 filesrc location=file.mp3 ! decodebin ! audioconvert ! "audio/x-raw-int,channels=2" ! deinterleave name=d interleave name=i ! audioconvert ! wavenc ! filesink location=test.wav d.src0 ! queue ! audioconvert ! i.sink1 d.src1 ! queue ! audioconvert ! i.sink0 - * ]| Decodes and deinterleaves a Stereo MP3 file into separate channels and - * then interleaves the channels again to a WAV file with the channel with the - * channels exchanged. - * |[ - * gst-launch-0.10 interleave name=i ! audioconvert ! wavenc ! filesink location=file.wav filesrc location=file1.wav ! decodebin ! audioconvert ! "audio/x-raw-int,channels=1" ! queue ! i.sink0 filesrc location=file2.wav ! decodebin ! audioconvert ! "audio/x-raw-int,channels=1" ! queue ! i.sink1 - * ]| Interleaves two Mono WAV files to a single Stereo WAV file. - * </refsect2> - */ - -#ifdef HAVE_CONFIG_H -# include "config.h" -#endif - -#include <gst/gst.h> -#include <string.h> -#include "interleave.h" - -#include <gst/audio/multichannel.h> - -GST_DEBUG_CATEGORY_STATIC (gst_interleave_debug); -#define GST_CAT_DEFAULT gst_interleave_debug - -static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink%d", - GST_PAD_SINK, - GST_PAD_REQUEST, - GST_STATIC_CAPS ("audio/x-raw-int, " - "rate = (int) [ 1, MAX ], " - "channels = (int) 1, " - "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " - "width = (int) { 8, 16, 24, 32 }, " - "depth = (int) [ 1, 32 ], " - "signed = (boolean) true; " - "audio/x-raw-float, " - "rate = (int) [ 1, MAX ], " - "channels = (int) 1, " - "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, " - "width = (int) { 32, 64 }") - ); - -static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, - GST_PAD_ALWAYS, - GST_STATIC_CAPS ("audio/x-raw-int, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, MAX ], " - "endianness = (int) { LITTLE_ENDIAN, BIG_ENDIAN }, " - "width = (int) { 8, 16, 24, 32 }, " - "depth = (int) [ 1, 32 ], " - "signed = (boolean) true; " - "audio/x-raw-float, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, MAX ], " - "endianness = (int) { LITTLE_ENDIAN , BIG_ENDIAN }, " - "width = (int) { 32, 64 }") - ); - -#define MAKE_FUNC(type) \ -static void interleave_##type (guint##type *out, guint##type *in, \ - guint stride, guint nframes) \ -{ \ - gint i; \ - \ - for (i = 0; i < nframes; i++) { \ - *out = in[i]; \ - out += stride; \ - } \ -} - -MAKE_FUNC (8); -MAKE_FUNC (16); -MAKE_FUNC (32); -MAKE_FUNC (64); - -static void -interleave_24 (guint8 * out, guint8 * in, guint stride, guint nframes) -{ - gint i; - - for (i = 0; i < nframes; i++) { - memcpy (out, in, 3); - out += stride * 3; - in += 3; - } -} - -typedef struct -{ - GstPad parent; - guint channel; -} GstInterleavePad; - -enum -{ - PROP_PAD_0, - PROP_PAD_CHANNEL -}; - -static void gst_interleave_pad_class_init (GstPadClass * klass); - -#define GST_TYPE_INTERLEAVE_PAD (gst_interleave_pad_get_type()) -#define GST_INTERLEAVE_PAD(pad) (G_TYPE_CHECK_INSTANCE_CAST((pad),GST_TYPE_INTERLEAVE_PAD,GstInterleavePad)) -#define GST_INTERLEAVE_PAD_CAST(pad) ((GstInterleavePad *) pad) -#define GST_IS_INTERLEAVE_PAD(pad) (G_TYPE_CHECK_INSTANCE_TYPE((pad),GST_TYPE_INTERLEAVE_PAD)) -static GType -gst_interleave_pad_get_type (void) -{ - static GType type = 0; - - if (G_UNLIKELY (type == 0)) { - type = g_type_register_static_simple (GST_TYPE_PAD, - g_intern_static_string ("GstInterleavePad"), sizeof (GstPadClass), - (GClassInitFunc) gst_interleave_pad_class_init, - sizeof (GstInterleavePad), NULL, 0); - } - return type; -} - -static void -gst_interleave_pad_get_property (GObject * object, - guint prop_id, GValue * value, GParamSpec * pspec) -{ - GstInterleavePad *self = GST_INTERLEAVE_PAD (object); - - switch (prop_id) { - case PROP_PAD_CHANNEL: - g_value_set_uint (value, self->channel); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_interleave_pad_class_init (GstPadClass * klass) -{ - GObjectClass *gobject_class = (GObjectClass *) klass; - - gobject_class->get_property = gst_interleave_pad_get_property; - - g_object_class_install_property (gobject_class, - PROP_PAD_CHANNEL, - g_param_spec_uint ("channel", - "Channel number", - "Number of the channel of this pad in the output", 0, G_MAXUINT, 0, - G_PARAM_READABLE | G_PARAM_STATIC_STRINGS)); -} - -GST_BOILERPLATE (GstInterleave, gst_interleave, GstElement, GST_TYPE_ELEMENT); - -enum -{ - PROP_0, - PROP_CHANNEL_POSITIONS, - PROP_CHANNEL_POSITIONS_FROM_INPUT -}; - -static void gst_interleave_set_property (GObject * object, - guint prop_id, const GValue * value, GParamSpec * pspec); -static void gst_interleave_get_property (GObject * object, - guint prop_id, GValue * value, GParamSpec * pspec); - -static GstPad *gst_interleave_request_new_pad (GstElement * element, - GstPadTemplate * templ, const gchar * name); -static void gst_interleave_release_pad (GstElement * element, GstPad * pad); - -static GstStateChangeReturn gst_interleave_change_state (GstElement * element, - GstStateChange transition); - -static gboolean gst_interleave_src_query (GstPad * pad, GstQuery * query); - -static gboolean gst_interleave_src_event (GstPad * pad, GstEvent * event); - -static gboolean gst_interleave_sink_event (GstPad * pad, GstEvent * event); - -static gboolean gst_interleave_sink_setcaps (GstPad * pad, GstCaps * caps); - -static GstCaps *gst_interleave_sink_getcaps (GstPad * pad); - -static GstFlowReturn gst_interleave_collected (GstCollectPads * pads, - GstInterleave * self); - -static void -gst_interleave_finalize (GObject * object) -{ - GstInterleave *self = GST_INTERLEAVE (object); - - if (self->collect) { - gst_object_unref (self->collect); - self->collect = NULL; - } - - if (self->channel_positions - && self->channel_positions != self->input_channel_positions) { - g_value_array_free (self->channel_positions); - self->channel_positions = NULL; - } - - if (self->input_channel_positions) { - g_value_array_free (self->input_channel_positions); - self->input_channel_positions = NULL; - } - - gst_caps_replace (&self->sinkcaps, NULL); - - G_OBJECT_CLASS (parent_class)->finalize (object); -} - -static gboolean -gst_interleave_check_channel_positions (GValueArray * positions) -{ - gint i; - - guint channels; - - GstAudioChannelPosition *pos; - - gboolean ret; - - channels = positions->n_values; - pos = g_new (GstAudioChannelPosition, positions->n_values); - - for (i = 0; i < channels; i++) { - GValue *v = g_value_array_get_nth (positions, i); - - pos[i] = g_value_get_enum (v); - } - - ret = gst_audio_check_channel_positions (pos, channels); - g_free (pos); - - return ret; -} - -static void -gst_interleave_set_channel_positions (GstInterleave * self, GstStructure * s) -{ - GValue pos_array = { 0, }; - gint i; - - g_value_init (&pos_array, GST_TYPE_ARRAY); - - if (self->channel_positions - && self->channels == self->channel_positions->n_values - && gst_interleave_check_channel_positions (self->channel_positions)) { - GST_DEBUG_OBJECT (self, "Using provided channel positions"); - for (i = 0; i < self->channels; i++) - gst_value_array_append_value (&pos_array, - g_value_array_get_nth (self->channel_positions, i)); - } else { - GValue pos_none = { 0, }; - - GST_WARNING_OBJECT (self, "Using NONE channel positions"); - - g_value_init (&pos_none, GST_TYPE_AUDIO_CHANNEL_POSITION); - g_value_set_enum (&pos_none, GST_AUDIO_CHANNEL_POSITION_NONE); - - for (i = 0; i < self->channels; i++) - gst_value_array_append_value (&pos_array, &pos_none); - - g_value_unset (&pos_none); - } - gst_structure_set_value (s, "channel-positions", &pos_array); - g_value_unset (&pos_array); -} - -static void -gst_interleave_base_init (gpointer g_class) -{ - gst_element_class_set_details_simple (g_class, "Audio interleaver", - "Filter/Converter/Audio", - "Folds many mono channels into one interleaved audio stream", - "Andy Wingo <wingo at pobox.com>, " - "Sebastian Dröge <slomo@circular-chaos.org>"); - - gst_element_class_add_pad_template (g_class, - gst_static_pad_template_get (&sink_template)); - gst_element_class_add_pad_template (g_class, - gst_static_pad_template_get (&src_template)); -} - -static void -gst_interleave_class_init (GstInterleaveClass * klass) -{ - GstElementClass *gstelement_class; - - GObjectClass *gobject_class; - - gobject_class = G_OBJECT_CLASS (klass); - gstelement_class = GST_ELEMENT_CLASS (klass); - - GST_DEBUG_CATEGORY_INIT (gst_interleave_debug, "interleave", 0, - "interleave element"); - - /* Reference GstInterleavePad class to have the type registered from - * a threadsafe context - */ - g_type_class_ref (GST_TYPE_INTERLEAVE_PAD); - - gobject_class->finalize = gst_interleave_finalize; - gobject_class->set_property = gst_interleave_set_property; - gobject_class->get_property = gst_interleave_get_property; - - /** - * GstInterleave:channel-positions - * - * Channel positions: This property controls the channel positions - * that are used on the src caps. The number of elements should be - * the same as the number of sink pads and the array should contain - * a valid list of channel positions. The n-th element of the array - * is the position of the n-th sink pad. - * - * These channel positions will only be used if they're valid and the - * number of elements is the same as the number of channels. If this - * is not given a NONE layout will be used. - * - */ - g_object_class_install_property (gobject_class, PROP_CHANNEL_POSITIONS, - g_param_spec_value_array ("channel-positions", "Channel positions", - "Channel positions used on the output", - g_param_spec_enum ("channel-position", "Channel position", - "Channel position of the n-th input", - GST_TYPE_AUDIO_CHANNEL_POSITION, - GST_AUDIO_CHANNEL_POSITION_NONE, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS), - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - /** - * GstInterleave:channel-positions-from-input - * - * Channel positions from input: If this property is set to %TRUE the channel - * positions will be taken from the input caps if valid channel positions for - * the output can be constructed from them. If this is set to %TRUE setting the - * channel-positions property overwrites this property again. - * - */ - g_object_class_install_property (gobject_class, - PROP_CHANNEL_POSITIONS_FROM_INPUT, - g_param_spec_boolean ("channel-positions-from-input", - "Channel positions from input", - "Take channel positions from the input", TRUE, - G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); - - gstelement_class->request_new_pad = - GST_DEBUG_FUNCPTR (gst_interleave_request_new_pad); - gstelement_class->release_pad = - GST_DEBUG_FUNCPTR (gst_interleave_release_pad); - gstelement_class->change_state = - GST_DEBUG_FUNCPTR (gst_interleave_change_state); -} - -static void -gst_interleave_init (GstInterleave * self, GstInterleaveClass * klass) -{ - self->src = gst_pad_new_from_static_template (&src_template, "src"); - - gst_pad_set_query_function (self->src, - GST_DEBUG_FUNCPTR (gst_interleave_src_query)); - gst_pad_set_event_function (self->src, - GST_DEBUG_FUNCPTR (gst_interleave_src_event)); - - gst_element_add_pad (GST_ELEMENT (self), self->src); - - self->collect = gst_collect_pads_new (); - gst_collect_pads_set_function (self->collect, - (GstCollectPadsFunction) gst_interleave_collected, self); - - self->input_channel_positions = g_value_array_new (0); - self->channel_positions_from_input = TRUE; - self->channel_positions = self->input_channel_positions; -} - -static void -gst_interleave_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstInterleave *self = GST_INTERLEAVE (object); - - switch (prop_id) { - case PROP_CHANNEL_POSITIONS: - if (self->channel_positions && - self->channel_positions != self->input_channel_positions) - g_value_array_free (self->channel_positions); - - self->channel_positions = g_value_dup_boxed (value); - self->channel_positions_from_input = FALSE; - break; - case PROP_CHANNEL_POSITIONS_FROM_INPUT: - self->channel_positions_from_input = g_value_get_boolean (value); - - if (self->channel_positions_from_input) { - if (self->channel_positions && - self->channel_positions != self->input_channel_positions) - g_value_array_free (self->channel_positions); - self->channel_positions = self->input_channel_positions; - } - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_interleave_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstInterleave *self = GST_INTERLEAVE (object); - - switch (prop_id) { - case PROP_CHANNEL_POSITIONS: - g_value_set_boxed (value, self->channel_positions); - break; - case PROP_CHANNEL_POSITIONS_FROM_INPUT: - g_value_set_boolean (value, self->channel_positions_from_input); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static GstPad * -gst_interleave_request_new_pad (GstElement * element, GstPadTemplate * templ, - const gchar * req_name) -{ - GstInterleave *self = GST_INTERLEAVE (element); - - GstPad *new_pad; - - gchar *pad_name; - - gint channels, padnumber; - GValue val = { 0, }; - - if (templ->direction != GST_PAD_SINK) - goto not_sink_pad; - - channels = g_atomic_int_exchange_and_add (&self->channels, 1); - padnumber = g_atomic_int_exchange_and_add (&self->padcounter, 1); - - pad_name = g_strdup_printf ("sink%d", padnumber); - new_pad = GST_PAD_CAST (g_object_new (GST_TYPE_INTERLEAVE_PAD, - "name", pad_name, "direction", templ->direction, - "template", templ, NULL)); - GST_INTERLEAVE_PAD_CAST (new_pad)->channel = channels; - GST_DEBUG_OBJECT (self, "requested new pad %s", pad_name); - g_free (pad_name); - - gst_pad_set_setcaps_function (new_pad, - GST_DEBUG_FUNCPTR (gst_interleave_sink_setcaps)); - gst_pad_set_getcaps_function (new_pad, - GST_DEBUG_FUNCPTR (gst_interleave_sink_getcaps)); - - gst_collect_pads_add_pad (self->collect, new_pad, sizeof (GstCollectData)); - - /* FIXME: hacked way to override/extend the event function of - * GstCollectPads; because it sets its own event function giving the - * element no access to events */ - self->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (new_pad); - gst_pad_set_event_function (new_pad, - GST_DEBUG_FUNCPTR (gst_interleave_sink_event)); - - if (!gst_element_add_pad (element, new_pad)) - goto could_not_add; - - g_value_init (&val, GST_TYPE_AUDIO_CHANNEL_POSITION); - g_value_set_enum (&val, GST_AUDIO_CHANNEL_POSITION_NONE); - self->input_channel_positions = - g_value_array_append (self->input_channel_positions, &val); - g_value_unset (&val); - - /* Update the src caps if we already have them */ - if (self->sinkcaps) { - GstCaps *srccaps; - - GstStructure *s; - - /* Take lock to make sure processing finishes first */ - GST_OBJECT_LOCK (self->collect); - - srccaps = gst_caps_copy (self->sinkcaps); - s = gst_caps_get_structure (srccaps, 0); - - gst_structure_set (s, "channels", G_TYPE_INT, self->channels, NULL); - gst_interleave_set_channel_positions (self, s); - - gst_pad_set_caps (self->src, srccaps); - gst_caps_unref (srccaps); - - GST_OBJECT_UNLOCK (self->collect); - } - - return new_pad; - - /* errors */ -not_sink_pad: - { - g_warning ("interleave: requested new pad that is not a SINK pad\n"); - return NULL; - } -could_not_add: - { - GST_DEBUG_OBJECT (self, "could not add pad %s", GST_PAD_NAME (new_pad)); - gst_collect_pads_remove_pad (self->collect, new_pad); - gst_object_unref (new_pad); - return NULL; - } -} - -static void -gst_interleave_release_pad (GstElement * element, GstPad * pad) -{ - GstInterleave *self = GST_INTERLEAVE (element); - - GList *l; - - g_return_if_fail (GST_IS_INTERLEAVE_PAD (pad)); - - /* Take lock to make sure we're not changing this when processing buffers */ - GST_OBJECT_LOCK (self->collect); - - g_atomic_int_add (&self->channels, -1); - - g_value_array_remove (self->input_channel_positions, - GST_INTERLEAVE_PAD_CAST (pad)->channel); - - /* Update channel numbers */ - GST_OBJECT_LOCK (self); - for (l = GST_ELEMENT_CAST (self)->sinkpads; l != NULL; l = l->next) { - GstInterleavePad *ipad = GST_INTERLEAVE_PAD (l->data); - - if (GST_INTERLEAVE_PAD_CAST (pad)->channel < ipad->channel) - ipad->channel--; - } - GST_OBJECT_UNLOCK (self); - - /* Update the src caps if we already have them */ - if (self->sinkcaps) { - if (self->channels > 0) { - GstCaps *srccaps; - - GstStructure *s; - - srccaps = gst_caps_copy (self->sinkcaps); - s = gst_caps_get_structure (srccaps, 0); - - gst_structure_set (s, "channels", G_TYPE_INT, self->channels, NULL); - gst_interleave_set_channel_positions (self, s); - - gst_pad_set_caps (self->src, srccaps); - gst_caps_unref (srccaps); - } else { - gst_caps_replace (&self->sinkcaps, NULL); - gst_pad_set_caps (self->src, NULL); - } - } - - GST_OBJECT_UNLOCK (self->collect); - - gst_collect_pads_remove_pad (self->collect, pad); - gst_element_remove_pad (element, pad); -} - -static GstStateChangeReturn -gst_interleave_change_state (GstElement * element, GstStateChange transition) -{ - GstInterleave *self; - - GstStateChangeReturn ret; - - self = GST_INTERLEAVE (element); - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - break; - case GST_STATE_CHANGE_READY_TO_PAUSED: - self->timestamp = 0; - self->offset = 0; - self->segment_pending = TRUE; - self->segment_position = 0; - self->segment_rate = 1.0; - gst_segment_init (&self->segment, GST_FORMAT_UNDEFINED); - gst_collect_pads_start (self->collect); - break; - case GST_STATE_CHANGE_PAUSED_TO_PLAYING: - break; - default: - break; - } - - /* Stop before calling the parent's state change function as - * GstCollectPads might take locks and we would deadlock in that - * case - */ - if (transition == GST_STATE_CHANGE_PAUSED_TO_READY) - gst_collect_pads_stop (self->collect); - - ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - switch (transition) { - case GST_STATE_CHANGE_PLAYING_TO_PAUSED: - break; - case GST_STATE_CHANGE_PAUSED_TO_READY: - gst_pad_set_caps (self->src, NULL); - gst_caps_replace (&self->sinkcaps, NULL); - break; - case GST_STATE_CHANGE_READY_TO_NULL: - break; - default: - break; - } - - return ret; -} - -static void -__remove_channels (GstCaps * caps) -{ - GstStructure *s; - - gint i, size; - - size = gst_caps_get_size (caps); - for (i = 0; i < size; i++) { - s = gst_caps_get_structure (caps, i); - gst_structure_remove_field (s, "channel-positions"); - gst_structure_remove_field (s, "channels"); - } -} - -static void -__set_channels (GstCaps * caps, gint channels) -{ - GstStructure *s; - - gint i, size; - - size = gst_caps_get_size (caps); - for (i = 0; i < size; i++) { - s = gst_caps_get_structure (caps, i); - if (channels > 0) - gst_structure_set (s, "channels", G_TYPE_INT, channels, NULL); - else - gst_structure_set (s, "channels", GST_TYPE_INT_RANGE, 1, G_MAXINT, NULL); - } -} - -/* we can only accept caps that we and downstream can handle. */ -static GstCaps * -gst_interleave_sink_getcaps (GstPad * pad) -{ - GstInterleave *self = GST_INTERLEAVE (gst_pad_get_parent (pad)); - - GstCaps *result, *peercaps, *sinkcaps; - - GST_OBJECT_LOCK (self); - - /* If we already have caps on one of the sink pads return them */ - if (self->sinkcaps) { - result = gst_caps_copy (self->sinkcaps); - } else { - /* get the downstream possible caps */ - peercaps = gst_pad_peer_get_caps (self->src); - /* get the allowed caps on this sinkpad */ - sinkcaps = gst_caps_copy (gst_pad_get_pad_template_caps (pad)); - __remove_channels (sinkcaps); - if (peercaps) { - __remove_channels (peercaps); - /* if the peer has caps, intersect */ - GST_DEBUG_OBJECT (pad, "intersecting peer and template caps"); - result = gst_caps_intersect (peercaps, sinkcaps); - gst_caps_unref (peercaps); - gst_caps_unref (sinkcaps); - } else { - /* the peer has no caps (or there is no peer), just use the allowed caps - * of this sinkpad. */ - GST_DEBUG_OBJECT (pad, "no peer caps, using sinkcaps"); - result = sinkcaps; - } - __set_channels (result, 1); - } - - GST_OBJECT_UNLOCK (self); - - gst_object_unref (self); - - GST_DEBUG_OBJECT (pad, "Returning caps %" GST_PTR_FORMAT, result); - - return result; -} - -static void -gst_interleave_set_process_function (GstInterleave * self) -{ - switch (self->width) { - case 8: - self->func = (GstInterleaveFunc) interleave_8; - break; - case 16: - self->func = (GstInterleaveFunc) interleave_16; - break; - case 24: - self->func = (GstInterleaveFunc) interleave_24; - break; - case 32: - self->func = (GstInterleaveFunc) interleave_32; - break; - case 64: - self->func = (GstInterleaveFunc) interleave_64; - break; - default: - g_assert_not_reached (); - break; - } -} - -static gboolean -gst_interleave_sink_setcaps (GstPad * pad, GstCaps * caps) -{ - GstInterleave *self; - - g_return_val_if_fail (GST_IS_INTERLEAVE_PAD (pad), FALSE); - - self = GST_INTERLEAVE (gst_pad_get_parent (pad)); - - /* First caps that are set on a sink pad are used as output caps */ - /* TODO: handle caps changes */ - if (self->sinkcaps && !gst_caps_is_subset (caps, self->sinkcaps)) { - goto cannot_change_caps; - } else { - GstCaps *srccaps; - - GstStructure *s; - - gboolean res; - - s = gst_caps_get_structure (caps, 0); - - if (!gst_structure_get_int (s, "width", &self->width)) - goto no_width; - - if (!gst_structure_get_int (s, "rate", &self->rate)) - goto no_rate; - - gst_interleave_set_process_function (self); - - if (gst_structure_has_field (s, "channel-positions")) { - const GValue *pos_array; - - pos_array = gst_structure_get_value (s, "channel-positions"); - if (GST_VALUE_HOLDS_ARRAY (pos_array) - && gst_value_array_get_size (pos_array) == 1) { - const GValue *pos = gst_value_array_get_value (pos_array, 0); - - GValue *apos = g_value_array_get_nth (self->input_channel_positions, - GST_INTERLEAVE_PAD_CAST (pad)->channel); - - g_value_set_enum (apos, g_value_get_enum (pos)); - } - } - - srccaps = gst_caps_copy (caps); - s = gst_caps_get_structure (srccaps, 0); - - gst_structure_set (s, "channels", G_TYPE_INT, self->channels, NULL); - gst_interleave_set_channel_positions (self, s); - - res = gst_pad_set_caps (self->src, srccaps); - gst_caps_unref (srccaps); - - if (!res) - goto src_did_not_accept; - } - - if (!self->sinkcaps) { - GstCaps *sinkcaps = gst_caps_copy (caps); - - GstStructure *s = gst_caps_get_structure (sinkcaps, 0); - - gst_structure_remove_field (s, "channel-positions"); - - gst_caps_replace (&self->sinkcaps, sinkcaps); - - gst_caps_unref (sinkcaps); - } - - gst_object_unref (self); - - return TRUE; - -cannot_change_caps: - { - GST_DEBUG_OBJECT (self, "caps of %" GST_PTR_FORMAT " already set, can't " - "change", self->sinkcaps); - gst_object_unref (self); - return FALSE; - } -src_did_not_accept: - { - GST_DEBUG_OBJECT (self, "src did not accept setcaps()"); - gst_object_unref (self); - return FALSE; - } -no_width: - { - GST_WARNING_OBJECT (self, "caps did not have width: %" GST_PTR_FORMAT, - caps); - gst_object_unref (self); - return FALSE; - } -no_rate: - { - GST_WARNING_OBJECT (self, "caps did not have rate: %" GST_PTR_FORMAT, caps); - gst_object_unref (self); - return FALSE; - } -} - -static gboolean -gst_interleave_sink_event (GstPad * pad, GstEvent * event) -{ - GstInterleave *self = GST_INTERLEAVE (gst_pad_get_parent (pad)); - - gboolean ret; - - GST_DEBUG ("Got %s event on pad %s:%s", GST_EVENT_TYPE_NAME (event), - GST_DEBUG_PAD_NAME (pad)); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_FLUSH_STOP: - /* mark a pending new segment. This event is synchronized - * with the streaming thread so we can safely update the - * variable without races. It's somewhat weird because we - * assume the collectpads forwarded the FLUSH_STOP past us - * and downstream (using our source pad, the bastard!). - */ - self->segment_pending = TRUE; - break; - default: - break; - } - - /* now GstCollectPads can take care of the rest, e.g. EOS */ - ret = self->collect_event (pad, event); - - gst_object_unref (self); - return ret; -} - -static gboolean -gst_interleave_src_query_duration (GstInterleave * self, GstQuery * query) -{ - gint64 max; - - gboolean res; - - GstFormat format; - - GstIterator *it; - - gboolean done; - - /* parse format */ - gst_query_parse_duration (query, &format, NULL); - - max = -1; - res = TRUE; - done = FALSE; - - /* Take maximum of all durations */ - it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (self)); - while (!done) { - GstIteratorResult ires; - - gpointer item; - - ires = gst_iterator_next (it, &item); - switch (ires) { - case GST_ITERATOR_DONE: - done = TRUE; - break; - case GST_ITERATOR_OK: - { - GstPad *pad = GST_PAD_CAST (item); - - gint64 duration; - - /* ask sink peer for duration */ - res &= gst_pad_query_peer_duration (pad, &format, &duration); - /* take max from all valid return values */ - if (res) { - /* valid unknown length, stop searching */ - if (duration == -1) { - max = duration; - done = TRUE; - } - /* else see if bigger than current max */ - else if (duration > max) - max = duration; - } - gst_object_unref (pad); - break; - } - case GST_ITERATOR_RESYNC: - max = -1; - res = TRUE; - gst_iterator_resync (it); - break; - default: - res = FALSE; - done = TRUE; - break; - } - } - gst_iterator_free (it); - - if (res) { - /* If in bytes format we have to multiply with the number of channels - * to get the correct results. All other formats should be fine */ - if (format == GST_FORMAT_BYTES && max != -1) - max *= self->channels; - - /* and store the max */ - GST_DEBUG_OBJECT (self, "Total duration in format %s: %" - GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max)); - gst_query_set_duration (query, format, max); - } - - return res; -} - -static gboolean -gst_interleave_src_query_latency (GstInterleave * self, GstQuery * query) -{ - GstClockTime min, max; - - gboolean live; - - gboolean res; - - GstIterator *it; - - gboolean done; - - res = TRUE; - done = FALSE; - - live = FALSE; - min = 0; - max = GST_CLOCK_TIME_NONE; - - /* Take maximum of all latency values */ - it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (self)); - while (!done) { - GstIteratorResult ires; - - gpointer item; - - ires = gst_iterator_next (it, &item); - switch (ires) { - case GST_ITERATOR_DONE: - done = TRUE; - break; - case GST_ITERATOR_OK: - { - GstPad *pad = GST_PAD_CAST (item); - - GstQuery *peerquery; - - GstClockTime min_cur, max_cur; - - gboolean live_cur; - - peerquery = gst_query_new_latency (); - - /* Ask peer for latency */ - res &= gst_pad_peer_query (pad, peerquery); - - /* take max from all valid return values */ - if (res) { - gst_query_parse_latency (peerquery, &live_cur, &min_cur, &max_cur); - - if (min_cur > min) - min = min_cur; - - if (max_cur != GST_CLOCK_TIME_NONE && - ((max != GST_CLOCK_TIME_NONE && max_cur > max) || - (max == GST_CLOCK_TIME_NONE))) - max = max_cur; - - live = live || live_cur; - } - - gst_query_unref (peerquery); - gst_object_unref (pad); - break; - } - case GST_ITERATOR_RESYNC: - live = FALSE; - min = 0; - max = GST_CLOCK_TIME_NONE; - res = TRUE; - gst_iterator_resync (it); - break; - default: - res = FALSE; - done = TRUE; - break; - } - } - gst_iterator_free (it); - - if (res) { - /* store the results */ - GST_DEBUG_OBJECT (self, "Calculated total latency: live %s, min %" - GST_TIME_FORMAT ", max %" GST_TIME_FORMAT, - (live ? "yes" : "no"), GST_TIME_ARGS (min), GST_TIME_ARGS (max)); - gst_query_set_latency (query, live, min, max); - } - - return res; -} - -static gboolean -gst_interleave_src_query (GstPad * pad, GstQuery * query) -{ - GstInterleave *self = GST_INTERLEAVE (gst_pad_get_parent (pad)); - - gboolean res = FALSE; - - switch (GST_QUERY_TYPE (query)) { - case GST_QUERY_POSITION: - { - GstFormat format; - - gst_query_parse_position (query, &format, NULL); - - switch (format) { - case GST_FORMAT_TIME: - /* FIXME, bring to stream time, might be tricky */ - gst_query_set_position (query, format, self->timestamp); - res = TRUE; - break; - case GST_FORMAT_BYTES: - gst_query_set_position (query, format, - self->offset * self->channels * self->width); - res = TRUE; - break; - case GST_FORMAT_DEFAULT: - gst_query_set_position (query, format, self->offset); - res = TRUE; - break; - default: - break; - } - break; - } - case GST_QUERY_DURATION: - res = gst_interleave_src_query_duration (self, query); - break; - case GST_QUERY_LATENCY: - res = gst_interleave_src_query_latency (self, query); - break; - default: - /* FIXME, needs a custom query handler because we have multiple - * sinkpads */ - res = gst_pad_query_default (pad, query); - break; - } - - gst_object_unref (self); - return res; -} - -static gboolean -forward_event_func (GstPad * pad, GValue * ret, GstEvent * event) -{ - gst_event_ref (event); - GST_LOG_OBJECT (pad, "About to send event %s", GST_EVENT_TYPE_NAME (event)); - if (!gst_pad_push_event (pad, event)) { - g_value_set_boolean (ret, FALSE); - GST_WARNING_OBJECT (pad, "Sending event %p (%s) failed.", - event, GST_EVENT_TYPE_NAME (event)); - } else { - GST_LOG_OBJECT (pad, "Sent event %p (%s).", - event, GST_EVENT_TYPE_NAME (event)); - } - gst_object_unref (pad); - return TRUE; -} - -static gboolean -forward_event (GstInterleave * self, GstEvent * event) -{ - gboolean ret; - - GstIterator *it; - GValue vret = { 0 }; - - GST_LOG_OBJECT (self, "Forwarding event %p (%s)", event, - GST_EVENT_TYPE_NAME (event)); - - ret = TRUE; - - g_value_init (&vret, G_TYPE_BOOLEAN); - g_value_set_boolean (&vret, TRUE); - it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (self)); - gst_iterator_fold (it, (GstIteratorFoldFunction) forward_event_func, &vret, - event); - gst_iterator_free (it); - gst_event_unref (event); - - ret = g_value_get_boolean (&vret); - - return ret; -} - - -static gboolean -gst_interleave_src_event (GstPad * pad, GstEvent * event) -{ - GstInterleave *self = GST_INTERLEAVE (gst_pad_get_parent (pad)); - - gboolean result; - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_QOS: - /* QoS might be tricky */ - result = FALSE; - break; - case GST_EVENT_SEEK: - { - GstSeekFlags flags; - - GstSeekType curtype; - - gint64 cur; - - /* parse the seek parameters */ - gst_event_parse_seek (event, &self->segment_rate, NULL, &flags, &curtype, - &cur, NULL, NULL); - - /* check if we are flushing */ - if (flags & GST_SEEK_FLAG_FLUSH) { - /* make sure we accept nothing anymore and return WRONG_STATE */ - gst_collect_pads_set_flushing (self->collect, TRUE); - - /* flushing seek, start flush downstream, the flush will be done - * when all pads received a FLUSH_STOP. */ - gst_pad_push_event (self->src, gst_event_new_flush_start ()); - } - - /* now wait for the collected to be finished and mark a new - * segment */ - GST_OBJECT_LOCK (self->collect); - if (curtype == GST_SEEK_TYPE_SET) - self->segment_position = cur; - else - self->segment_position = 0; - self->segment_pending = TRUE; - GST_OBJECT_UNLOCK (self->collect); - - result = forward_event (self, event); - break; - } - case GST_EVENT_NAVIGATION: - /* navigation is rather pointless. */ - result = FALSE; - break; - default: - /* just forward the rest for now */ - result = forward_event (self, event); - break; - } - gst_object_unref (self); - - return result; -} - -static GstFlowReturn -gst_interleave_collected (GstCollectPads * pads, GstInterleave * self) -{ - guint size; - - GstBuffer *outbuf; - - GstFlowReturn ret = GST_FLOW_OK; - - GSList *collected; - - guint nsamples; - - guint ncollected = 0; - - gboolean empty = TRUE; - - gint width = self->width / 8; - - g_return_val_if_fail (self->func != NULL, GST_FLOW_NOT_NEGOTIATED); - g_return_val_if_fail (self->width > 0, GST_FLOW_NOT_NEGOTIATED); - g_return_val_if_fail (self->channels > 0, GST_FLOW_NOT_NEGOTIATED); - g_return_val_if_fail (self->rate > 0, GST_FLOW_NOT_NEGOTIATED); - - size = gst_collect_pads_available (pads); - - g_return_val_if_fail (size % width == 0, GST_FLOW_ERROR); - - GST_DEBUG_OBJECT (self, "Starting to collect %u bytes from %d channels", size, - self->channels); - - nsamples = size / width; - - ret = - gst_pad_alloc_buffer (self->src, GST_BUFFER_OFFSET_NONE, - size * self->channels, GST_PAD_CAPS (self->src), &outbuf); - - if (ret != GST_FLOW_OK) { - return ret; - } else if (outbuf == NULL || GST_BUFFER_SIZE (outbuf) < size * self->channels) { - gst_buffer_unref (outbuf); - return GST_FLOW_NOT_NEGOTIATED; - } else if (!gst_caps_is_equal (GST_BUFFER_CAPS (outbuf), - GST_PAD_CAPS (self->src))) { - gst_buffer_unref (outbuf); - return GST_FLOW_NOT_NEGOTIATED; - } - - memset (GST_BUFFER_DATA (outbuf), 0, size * self->channels); - - for (collected = pads->data; collected != NULL; collected = collected->next) { - GstCollectData *cdata; - - GstBuffer *inbuf; - - guint8 *outdata; - - cdata = (GstCollectData *) collected->data; - - inbuf = gst_collect_pads_take_buffer (pads, cdata, size); - if (inbuf == NULL) { - GST_DEBUG_OBJECT (cdata->pad, "No buffer available"); - goto next; - } - ncollected++; - - if (GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP)) - goto next; - - empty = FALSE; - outdata = - GST_BUFFER_DATA (outbuf) + - width * GST_INTERLEAVE_PAD_CAST (cdata->pad)->channel; - - self->func (outdata, GST_BUFFER_DATA (inbuf), self->channels, nsamples); - - next: - if (inbuf) - gst_buffer_unref (inbuf); - } - - if (ncollected == 0) - goto eos; - - if (self->segment_pending) { - GstEvent *event; - - event = gst_event_new_new_segment_full (FALSE, self->segment_rate, - 1.0, GST_FORMAT_TIME, self->timestamp, -1, self->segment_position); - - gst_pad_push_event (self->src, event); - self->segment_pending = FALSE; - self->segment_position = 0; - } - - GST_BUFFER_TIMESTAMP (outbuf) = self->timestamp; - GST_BUFFER_OFFSET (outbuf) = self->offset; - - self->offset += nsamples; - self->timestamp = gst_util_uint64_scale_int (self->offset, - GST_SECOND, self->rate); - - GST_BUFFER_DURATION (outbuf) = self->timestamp - - GST_BUFFER_TIMESTAMP (outbuf); - - if (empty) - GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_GAP); - - GST_LOG_OBJECT (self, "pushing outbuf, timestamp %" GST_TIME_FORMAT, - GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf))); - ret = gst_pad_push (self->src, outbuf); - - return ret; - -eos: - { - GST_DEBUG_OBJECT (self, "no data available, must be EOS"); - gst_buffer_unref (outbuf); - gst_pad_push_event (self->src, gst_event_new_eos ()); - return GST_FLOW_UNEXPECTED; - } -} diff --git a/gst/interleave/interleave.h b/gst/interleave/interleave.h deleted file mode 100644 index fb3b2741..00000000 --- a/gst/interleave/interleave.h +++ /dev/null @@ -1,89 +0,0 @@ -/* GStreamer - * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> - * 2000 Wim Taymans <wtay@chello.be> - * 2005 Wim Taymans <wim@fluendo.com> - * 2007 Andy Wingo <wingo at pobox.com> - * 2008 Sebastian Dröge <slomo@circular-chaos.org> - * - * interleave.c: interleave samples, mostly based on adder - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifndef __INTERLEAVE_H__ -#define __INTERLEAVE_H__ - -#include <gst/gst.h> -#include <gst/base/gstcollectpads.h> - -G_BEGIN_DECLS - -#define GST_TYPE_INTERLEAVE (gst_interleave_get_type()) -#define GST_INTERLEAVE(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_INTERLEAVE,GstInterleave)) -#define GST_INTERLEAVE_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_INTERLEAVE,GstInterleaveClass)) -#define GST_INTERLEAVE_GET_CLASS(obj) \ - (G_TYPE_INSTANCE_GET_CLASS ((obj),GST_TYPE_INTERLEAVE,GstInterleaveClass)) -#define GST_IS_INTERLEAVE(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_INTERLEAVE)) -#define GST_IS_INTERLEAVE_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_INTERLEAVE)) - -typedef struct _GstInterleave GstInterleave; -typedef struct _GstInterleaveClass GstInterleaveClass; - -typedef void (*GstInterleaveFunc) (gpointer out, gpointer in, guint stride, guint nframes); - -struct _GstInterleave -{ - GstElement element; - - /*< private >*/ - GstCollectPads *collect; - - gint channels; - gint padcounter; - gint rate; - gint width; - - GValueArray *channel_positions; - GValueArray *input_channel_positions; - gboolean channel_positions_from_input; - - GstCaps *sinkcaps; - - GstClockTime timestamp; - guint64 offset; - - gboolean segment_pending; - guint64 segment_position; - gdouble segment_rate; - GstSegment segment; - - GstPadEventFunction collect_event; - - GstInterleaveFunc func; - - GstPad *src; -}; - -struct _GstInterleaveClass -{ - GstElementClass parent_class; -}; - -GType gst_interleave_get_type (void); - -G_END_DECLS - -#endif /* __INTERLEAVE_H__ */ diff --git a/gst/interleave/plugin.c b/gst/interleave/plugin.c deleted file mode 100644 index 7017c45c..00000000 --- a/gst/interleave/plugin.c +++ /dev/null @@ -1,44 +0,0 @@ -/* GStreamer interleave plugin - * Copyright (C) 2004,2007 Andy Wingo <wingo at pobox.com> - * - * plugin.c: the stubs for the interleave plugin - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - -#ifdef HAVE_CONFIG_H -#include "config.h" -#endif - -#include "plugin.h" - -static gboolean -plugin_init (GstPlugin * plugin) -{ - if (!gst_element_register (plugin, "interleave", - GST_RANK_NONE, gst_interleave_get_type ()) || - !gst_element_register (plugin, "deinterleave", - GST_RANK_NONE, gst_deinterleave_get_type ())) - return FALSE; - - return TRUE; -} - -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, - GST_VERSION_MINOR, - "interleave", - "Audio interleaver/deinterleaver", - plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN); diff --git a/gst/interleave/plugin.h b/gst/interleave/plugin.h deleted file mode 100644 index 3e96a7e1..00000000 --- a/gst/interleave/plugin.h +++ /dev/null @@ -1,31 +0,0 @@ -/* GStreamer interleave plugin - * Copyright (C) 2004,2007 Andy Wingo <wingo at pobox.com> - * - * plugin.h: the stubs for the interleave plugin - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Library General Public - * License as published by the Free Software Foundation; either - * version 2 of the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, - * but WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Library General Public License for more details. - * - * You should have received a copy of the GNU Library General Public - * License along with this library; if not, write to the - * Free Software Foundation, Inc., 59 Temple Place - Suite 330, - * Boston, MA 02111-1307, USA. - */ - - -#ifndef __GST_PLUGIN_INTERLEAVE_H__ -#define __GST_PLUGIN_INTERLEAVE_H__ - - -#include <gst/gst.h> -#include "interleave.h" -#include "deinterleave.h" - -#endif /* __GST_PLUGIN_INTERLEAVE_H__ */ diff --git a/gst/replaygain/Makefile.am b/gst/replaygain/Makefile.am deleted file mode 100644 index a0a3ca5a..00000000 --- a/gst/replaygain/Makefile.am +++ /dev/null @@ -1,21 +0,0 @@ -plugin_LTLIBRARIES = libgstreplaygain.la - -libgstreplaygain_la_SOURCES = \ - gstrganalysis.c \ - gstrglimiter.c \ - gstrgvolume.c \ - replaygain.c \ - rganalysis.c -libgstreplaygain_la_CFLAGS = \ - $(GST_CFLAGS) $(GST_BASE_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS) -libgstreplaygain_la_LIBADD = \ - $(GST_LIBS) $(GST_BASE_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstpbutils-0.10 $(LIBM) -libgstreplaygain_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) - -# headers we need but don't want installed -noinst_HEADERS = \ - gstrganalysis.h \ - gstrglimiter.h \ - gstrgvolume.h \ - replaygain.h \ - rganalysis.h diff --git a/gst/replaygain/gstrganalysis.c b/gst/replaygain/gstrganalysis.c deleted file mode 100644 index 982c8a7f..00000000 --- a/gst/replaygain/gstrganalysis.c +++ /dev/null @@ -1,692 +0,0 @@ -/* GStreamer ReplayGain analysis - * - * Copyright (C) 2006 Rene Stadler <mail@renestadler.de> - * - * gstrganalysis.c: Element that performs the ReplayGain analysis - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -/** - * SECTION:element-rganalysis - * @see_also: #GstRgVolume - * - * This element analyzes raw audio sample data in accordance with the proposed - * <ulink url="http://replaygain.org">ReplayGain standard</ulink> for - * calculating the ideal replay gain for music tracks and albums. The element - * is designed as a pass-through filter that never modifies any data. As it - * receives an EOS event, it finalizes the ongoing analysis and generates a tag - * list containing the results. It is sent downstream with a tag event and - * posted on the message bus with a tag message. The EOS event is forwarded as - * normal afterwards. Result tag lists at least contain the tags - * #GST_TAG_TRACK_GAIN, #GST_TAG_TRACK_PEAK and #GST_TAG_REFERENCE_LEVEL. - * - * Because the generated metadata tags become available at the end of streams, - * downstream muxer and encoder elements are normally unable to save them in - * their output since they generally save metadata in the file header. - * Therefore, it is often necessary that applications read the results in a bus - * event handler for the tag message. Obtaining the values this way is always - * needed for <link linkend="GstRgAnalysis--num-tracks">album processing</link> - * since the album gain and peak values need to be associated with all tracks of - * an album, not just the last one. - * - * <refsect2> - * <title>Example launch lines</title> - * |[ - * gst-launch -t audiotestsrc wave=sine num-buffers=512 ! rganalysis ! fakesink - * ]| Analyze a simple test waveform - * |[ - * gst-launch -t filesrc location=filename.ext ! decodebin \ - * ! audioconvert ! audioresample ! rganalysis ! fakesink - * ]| Analyze a given file - * |[ - * gst-launch -t gnomevfssrc location=http://replaygain.hydrogenaudio.org/ref_pink.wav \ - * ! wavparse ! rganalysis ! fakesink - * ]| Analyze the pink noise reference file - * <para> - * The above launch line yields a result gain of +6 dB (instead of the expected - * +0 dB). This is not in error, refer to the #GstRgAnalysis:reference-level - * property documentation for more information. - * </para> - * </refsect2> - * <refsect2> - * <title>Acknowledgements</title> - * <para> - * This element is based on code used in the <ulink - * url="http://sjeng.org/vorbisgain.html">vorbisgain</ulink> program and many - * others. The relevant parts are copyrighted by David Robinson, Glen Sawyer - * and Frank Klemm. - * </para> - * </refsect2> - */ - -#ifdef HAVE_CONFIG_H -#include <config.h> -#endif - -#include <gst/gst.h> -#include <gst/base/gstbasetransform.h> - -#include "gstrganalysis.h" -#include "replaygain.h" - -GST_DEBUG_CATEGORY_STATIC (gst_rg_analysis_debug); -#define GST_CAT_DEFAULT gst_rg_analysis_debug - -static const GstElementDetails rganalysis_details = { - "ReplayGain analysis", - "Filter/Analyzer/Audio", - "Perform the ReplayGain analysis", - "Ren\xc3\xa9 Stadler <mail@renestadler.de>" -}; - -/* Default property value. */ -#define FORCED_DEFAULT TRUE - -enum -{ - PROP_0, - PROP_NUM_TRACKS, - PROP_FORCED, - PROP_REFERENCE_LEVEL -}; - -/* The ReplayGain algorithm is intended for use with mono and stereo - * audio. The used implementation has filter coefficients for the - * "usual" sample rates in the 8000 to 48000 Hz range. */ -#define REPLAY_GAIN_CAPS \ - "channels = (int) { 1, 2 }, " \ - "rate = (int) { 8000, 11025, 12000, 16000, 22050, 24000, 32000, " \ - "44100, 48000 }" - -static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "width = (int) 32, " "endianness = (int) BYTE_ORDER, " - REPLAY_GAIN_CAPS "; " - "audio/x-raw-int, " - "width = (int) 16, " "depth = (int) [ 1, 16 ], " - "signed = (boolean) true, " "endianness = (int) BYTE_ORDER, " - REPLAY_GAIN_CAPS)); - -static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "width = (int) 32, " "endianness = (int) BYTE_ORDER, " - REPLAY_GAIN_CAPS "; " - "audio/x-raw-int, " - "width = (int) 16, " "depth = (int) [ 1, 16 ], " - "signed = (boolean) true, " "endianness = (int) BYTE_ORDER, " - REPLAY_GAIN_CAPS)); - -GST_BOILERPLATE (GstRgAnalysis, gst_rg_analysis, GstBaseTransform, - GST_TYPE_BASE_TRANSFORM); - -static void gst_rg_analysis_class_init (GstRgAnalysisClass * klass); -static void gst_rg_analysis_init (GstRgAnalysis * filter, - GstRgAnalysisClass * gclass); - -static void gst_rg_analysis_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_rg_analysis_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static gboolean gst_rg_analysis_start (GstBaseTransform * base); -static gboolean gst_rg_analysis_set_caps (GstBaseTransform * base, - GstCaps * incaps, GstCaps * outcaps); -static GstFlowReturn gst_rg_analysis_transform_ip (GstBaseTransform * base, - GstBuffer * buf); -static gboolean gst_rg_analysis_event (GstBaseTransform * base, - GstEvent * event); -static gboolean gst_rg_analysis_stop (GstBaseTransform * base); - -static void gst_rg_analysis_handle_tags (GstRgAnalysis * filter, - const GstTagList * tag_list); -static void gst_rg_analysis_handle_eos (GstRgAnalysis * filter); -static gboolean gst_rg_analysis_track_result (GstRgAnalysis * filter, - GstTagList ** tag_list); -static gboolean gst_rg_analysis_album_result (GstRgAnalysis * filter, - GstTagList ** tag_list); - -static void -gst_rg_analysis_base_init (gpointer g_class) -{ - GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_factory)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_factory)); - gst_element_class_set_details (element_class, &rganalysis_details); - - GST_DEBUG_CATEGORY_INIT (gst_rg_analysis_debug, "rganalysis", 0, - "ReplayGain analysis element"); -} - -static void -gst_rg_analysis_class_init (GstRgAnalysisClass * klass) -{ - GObjectClass *gobject_class; - GstBaseTransformClass *trans_class; - - gobject_class = (GObjectClass *) klass; - gobject_class->set_property = gst_rg_analysis_set_property; - gobject_class->get_property = gst_rg_analysis_get_property; - - /** - * GstRgAnalysis:num-tracks: - * - * Number of remaining album tracks. - * - * Analyzing several streams sequentially and assigning them a common result - * gain is known as "album processing". If this gain is used during playback - * (by switching to "album mode"), all tracks of an album receive the same - * amplification. This keeps the relative volume levels between the tracks - * intact. To enable this, set this property to the number of streams that - * will be processed as album tracks. - * - * Every time an EOS event is received, the value of this property is - * decremented by one. As it reaches zero, it is assumed that the last track - * of the album finished. The tag list for the final stream will contain the - * additional tags #GST_TAG_ALBUM_GAIN and #GST_TAG_ALBUM_PEAK. All other - * streams just get the two track tags posted because the values for the album - * tags are not known before all tracks are analyzed. Applications need to - * ensure that the album gain and peak values are also associated with the - * other tracks when storing the results. - * - * If the total number of album tracks is unknown beforehand, just ensure that - * the value is greater than 1 before each track starts. Then before the end - * of the last track, set it to the value 1. - * - * To perform album processing, the element has to preserve data between - * streams. This cannot survive a state change to the NULL or READY state. - * If you change your pipeline's state to NULL or READY between tracks, lock - * the element's state using gst_element_set_locked_state() when it is in - * PAUSED or PLAYING. - */ - g_object_class_install_property (gobject_class, PROP_NUM_TRACKS, - g_param_spec_int ("num-tracks", "Number of album tracks", - "Number of remaining album tracks", 0, G_MAXINT, 0, - G_PARAM_READWRITE)); - /** - * GstRgAnalysis:forced: - * - * Whether to analyze streams even when ReplayGain tags exist. - * - * For assisting transcoder/converter applications, the element can silently - * skip the processing of streams that already contain the necessary tags. - * Data will flow as usual but the element will not consume CPU time and will - * not generate result tags. To enable possible skipping, set this property - * to #FALSE. - * - * If used in conjunction with <link linkend="GstRgAnalysis--num-tracks">album - * processing</link>, the element will skip the number of remaining album - * tracks if a full set of tags is found for the first track. If a subsequent - * track of the album is missing tags, processing cannot start again. If this - * is undesired, the application has to scan all files beforehand and enable - * forcing of processing if needed. - */ - g_object_class_install_property (gobject_class, PROP_FORCED, - g_param_spec_boolean ("forced", "Forced", - "Analyze even if ReplayGain tags exist", - FORCED_DEFAULT, G_PARAM_READWRITE)); - /** - * GstRgAnalysis:reference-level: - * - * Reference level [dB]. - * - * Analyzing the ReplayGain pink noise reference waveform computes a result of - * +6 dB instead of the expected 0 dB. This is because the default reference - * level is 89 dB. To obtain values as lined out in the original proposal of - * ReplayGain, set this property to 83. - * - * Almost all software uses 89 dB as a reference however, and this value has - * become the new official value. That is to say, while the change has been - * acclaimed by the author of the ReplayGain proposal, the <ulink - * url="http://replaygain.org">webpage</ulink> is still outdated at the time - * of this writing. - * - * The value was changed because the original proposal recommends a default - * pre-amp value of +6 dB for playback. This seemed a bit odd, as it means - * that the algorithm has the general tendency to produce adjustment values - * that are 6 dB too low. Bumping the reference level by 6 dB compensated for - * this. - * - * The problem of the reference level being ambiguous for lack of concise - * standardization is to be solved by adopting the #GST_TAG_REFERENCE_LEVEL - * tag, which allows to store the used value alongside the gain values. - */ - g_object_class_install_property (gobject_class, PROP_REFERENCE_LEVEL, - g_param_spec_double ("reference-level", "Reference level", - "Reference level [dB]", 0.0, 150., RG_REFERENCE_LEVEL, - G_PARAM_READWRITE)); - - trans_class = (GstBaseTransformClass *) klass; - trans_class->start = GST_DEBUG_FUNCPTR (gst_rg_analysis_start); - trans_class->set_caps = GST_DEBUG_FUNCPTR (gst_rg_analysis_set_caps); - trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_analysis_transform_ip); - trans_class->event = GST_DEBUG_FUNCPTR (gst_rg_analysis_event); - trans_class->stop = GST_DEBUG_FUNCPTR (gst_rg_analysis_stop); - trans_class->passthrough_on_same_caps = TRUE; -} - -static void -gst_rg_analysis_init (GstRgAnalysis * filter, GstRgAnalysisClass * gclass) -{ - GstBaseTransform *base = GST_BASE_TRANSFORM (filter); - - gst_base_transform_set_gap_aware (base, TRUE); - - filter->num_tracks = 0; - filter->forced = FORCED_DEFAULT; - filter->reference_level = RG_REFERENCE_LEVEL; - - filter->ctx = NULL; - filter->analyze = NULL; -} - -static void -gst_rg_analysis_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstRgAnalysis *filter = GST_RG_ANALYSIS (object); - - switch (prop_id) { - case PROP_NUM_TRACKS: - filter->num_tracks = g_value_get_int (value); - break; - case PROP_FORCED: - filter->forced = g_value_get_boolean (value); - break; - case PROP_REFERENCE_LEVEL: - filter->reference_level = g_value_get_double (value); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_rg_analysis_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstRgAnalysis *filter = GST_RG_ANALYSIS (object); - - switch (prop_id) { - case PROP_NUM_TRACKS: - g_value_set_int (value, filter->num_tracks); - break; - case PROP_FORCED: - g_value_set_boolean (value, filter->forced); - break; - case PROP_REFERENCE_LEVEL: - g_value_set_double (value, filter->reference_level); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static gboolean -gst_rg_analysis_start (GstBaseTransform * base) -{ - GstRgAnalysis *filter = GST_RG_ANALYSIS (base); - - filter->ignore_tags = FALSE; - filter->skip = FALSE; - filter->has_track_gain = FALSE; - filter->has_track_peak = FALSE; - filter->has_album_gain = FALSE; - filter->has_album_peak = FALSE; - - filter->ctx = rg_analysis_new (); - filter->analyze = NULL; - - GST_LOG_OBJECT (filter, "started"); - - return TRUE; -} - -static gboolean -gst_rg_analysis_set_caps (GstBaseTransform * base, GstCaps * in_caps, - GstCaps * out_caps) -{ - GstRgAnalysis *filter = GST_RG_ANALYSIS (base); - GstStructure *structure; - const gchar *name; - gint n_channels, sample_rate, sample_bit_size, sample_size; - - g_return_val_if_fail (filter->ctx != NULL, FALSE); - - GST_DEBUG_OBJECT (filter, - "set_caps in %" GST_PTR_FORMAT " out %" GST_PTR_FORMAT, - in_caps, out_caps); - - structure = gst_caps_get_structure (in_caps, 0); - name = gst_structure_get_name (structure); - - if (!gst_structure_get_int (structure, "width", &sample_bit_size) - || !gst_structure_get_int (structure, "channels", &n_channels) - || !gst_structure_get_int (structure, "rate", &sample_rate)) - goto invalid_format; - - if (!rg_analysis_set_sample_rate (filter->ctx, sample_rate)) - goto invalid_format; - - if (sample_bit_size % 8 != 0) - goto invalid_format; - sample_size = sample_bit_size / 8; - - if (g_str_equal (name, "audio/x-raw-float")) { - - if (sample_size != sizeof (gfloat)) - goto invalid_format; - - /* The depth is not variable for float formats of course. It just - * makes the transform function nice and simple if the - * rg_analysis_analyze_* functions have a common signature. */ - filter->depth = sizeof (gfloat) * 8; - - if (n_channels == 1) - filter->analyze = rg_analysis_analyze_mono_float; - else if (n_channels == 2) - filter->analyze = rg_analysis_analyze_stereo_float; - else - goto invalid_format; - - } else if (g_str_equal (name, "audio/x-raw-int")) { - - if (sample_size != sizeof (gint16)) - goto invalid_format; - - if (!gst_structure_get_int (structure, "depth", &filter->depth)) - goto invalid_format; - if (filter->depth < 1 || filter->depth > 16) - goto invalid_format; - - if (n_channels == 1) - filter->analyze = rg_analysis_analyze_mono_int16; - else if (n_channels == 2) - filter->analyze = rg_analysis_analyze_stereo_int16; - else - goto invalid_format; - - } else { - - goto invalid_format; - } - - return TRUE; - - /* Errors. */ -invalid_format: - { - filter->analyze = NULL; - GST_ELEMENT_ERROR (filter, CORE, NEGOTIATION, - ("Invalid incoming caps: %" GST_PTR_FORMAT, in_caps), (NULL)); - return FALSE; - } -} - -static GstFlowReturn -gst_rg_analysis_transform_ip (GstBaseTransform * base, GstBuffer * buf) -{ - GstRgAnalysis *filter = GST_RG_ANALYSIS (base); - - g_return_val_if_fail (filter->ctx != NULL, GST_FLOW_WRONG_STATE); - g_return_val_if_fail (filter->analyze != NULL, GST_FLOW_NOT_NEGOTIATED); - - if (filter->skip) - return GST_FLOW_OK; - - /* Buffers made up of silence have no influence on the analysis: */ - if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)) - return GST_FLOW_OK; - - GST_LOG_OBJECT (filter, "processing buffer of size %u", - GST_BUFFER_SIZE (buf)); - - filter->analyze (filter->ctx, GST_BUFFER_DATA (buf), GST_BUFFER_SIZE (buf), - filter->depth); - - return GST_FLOW_OK; -} - -static gboolean -gst_rg_analysis_event (GstBaseTransform * base, GstEvent * event) -{ - GstRgAnalysis *filter = GST_RG_ANALYSIS (base); - - g_return_val_if_fail (filter->ctx != NULL, TRUE); - - switch (GST_EVENT_TYPE (event)) { - - case GST_EVENT_EOS: - { - GST_LOG_OBJECT (filter, "received EOS event"); - - gst_rg_analysis_handle_eos (filter); - - GST_LOG_OBJECT (filter, "passing on EOS event"); - - break; - } - case GST_EVENT_TAG: - { - GstTagList *tag_list; - - /* The reference to the tag list is borrowed. */ - gst_event_parse_tag (event, &tag_list); - gst_rg_analysis_handle_tags (filter, tag_list); - - break; - } - default: - break; - } - - return GST_BASE_TRANSFORM_CLASS (parent_class)->event (base, event); -} - -static gboolean -gst_rg_analysis_stop (GstBaseTransform * base) -{ - GstRgAnalysis *filter = GST_RG_ANALYSIS (base); - - g_return_val_if_fail (filter->ctx != NULL, FALSE); - - rg_analysis_destroy (filter->ctx); - filter->ctx = NULL; - - GST_LOG_OBJECT (filter, "stopped"); - - return TRUE; -} - -static void -gst_rg_analysis_handle_tags (GstRgAnalysis * filter, - const GstTagList * tag_list) -{ - gboolean album_processing = (filter->num_tracks > 0); - gdouble dummy; - - if (!album_processing) - filter->ignore_tags = FALSE; - - if (filter->skip && album_processing) { - GST_DEBUG_OBJECT (filter, "ignoring tag event: skipping album"); - return; - } else if (filter->skip) { - GST_DEBUG_OBJECT (filter, "ignoring tag event: skipping track"); - return; - } else if (filter->ignore_tags) { - GST_DEBUG_OBJECT (filter, "ignoring tag event: cannot skip anyways"); - return; - } - - filter->has_track_gain |= gst_tag_list_get_double (tag_list, - GST_TAG_TRACK_GAIN, &dummy); - filter->has_track_peak |= gst_tag_list_get_double (tag_list, - GST_TAG_TRACK_PEAK, &dummy); - filter->has_album_gain |= gst_tag_list_get_double (tag_list, - GST_TAG_ALBUM_GAIN, &dummy); - filter->has_album_peak |= gst_tag_list_get_double (tag_list, - GST_TAG_ALBUM_PEAK, &dummy); - - if (!(filter->has_track_gain && filter->has_track_peak)) { - GST_DEBUG_OBJECT (filter, "track tags not complete yet"); - return; - } - - if (album_processing && !(filter->has_album_gain && filter->has_album_peak)) { - GST_DEBUG_OBJECT (filter, "album tags not complete yet"); - return; - } - - if (filter->forced) { - GST_DEBUG_OBJECT (filter, - "existing tags are sufficient, but processing anyway (forced)"); - return; - } - - filter->skip = TRUE; - rg_analysis_reset (filter->ctx); - - if (!album_processing) { - GST_DEBUG_OBJECT (filter, - "existing tags are sufficient, will not process this track"); - } else { - GST_DEBUG_OBJECT (filter, - "existing tags are sufficient, will not process this album"); - } -} - -static void -gst_rg_analysis_handle_eos (GstRgAnalysis * filter) -{ - gboolean album_processing = (filter->num_tracks > 0); - gboolean album_finished = (filter->num_tracks == 1); - gboolean album_skipping = album_processing && filter->skip; - - filter->has_track_gain = FALSE; - filter->has_track_peak = FALSE; - - if (album_finished) { - filter->ignore_tags = FALSE; - filter->skip = FALSE; - filter->has_album_gain = FALSE; - filter->has_album_peak = FALSE; - } else if (!album_skipping) { - filter->skip = FALSE; - } - - /* We might have just fully processed a track because it has - * incomplete tags. If we do album processing and allow skipping - * (not forced), prevent switching to skipping if a later track with - * full tags comes along: */ - if (!filter->forced && album_processing && !album_finished) - filter->ignore_tags = TRUE; - - if (!filter->skip) { - GstTagList *tag_list = NULL; - gboolean track_success; - gboolean album_success = FALSE; - - track_success = gst_rg_analysis_track_result (filter, &tag_list); - - if (album_finished) - album_success = gst_rg_analysis_album_result (filter, &tag_list); - else if (!album_processing) - rg_analysis_reset_album (filter->ctx); - - if (track_success || album_success) { - GST_LOG_OBJECT (filter, "posting tag list with results"); - gst_tag_list_add (tag_list, GST_TAG_MERGE_APPEND, - GST_TAG_REFERENCE_LEVEL, filter->reference_level, NULL); - /* This steals our reference to the list: */ - gst_element_found_tags_for_pad (GST_ELEMENT (filter), - GST_BASE_TRANSFORM_SRC_PAD (GST_BASE_TRANSFORM (filter)), tag_list); - } - } - - if (album_processing) { - filter->num_tracks--; - - if (!album_finished) { - GST_DEBUG_OBJECT (filter, "album not finished yet (num-tracks is now %u)", - filter->num_tracks); - } else { - GST_DEBUG_OBJECT (filter, "album finished (num-tracks is now 0)"); - } - } - - if (album_processing) - g_object_notify (G_OBJECT (filter), "num-tracks"); -} - -static gboolean -gst_rg_analysis_track_result (GstRgAnalysis * filter, GstTagList ** tag_list) -{ - gboolean track_success; - gdouble track_gain, track_peak; - - track_success = rg_analysis_track_result (filter->ctx, &track_gain, - &track_peak); - - if (track_success) { - track_gain += filter->reference_level - RG_REFERENCE_LEVEL; - GST_INFO_OBJECT (filter, "track gain is %+.2f dB, peak %.6f", track_gain, - track_peak); - } else { - GST_INFO_OBJECT (filter, "track was too short to analyze"); - } - - if (track_success) { - if (*tag_list == NULL) - *tag_list = gst_tag_list_new (); - gst_tag_list_add (*tag_list, GST_TAG_MERGE_APPEND, - GST_TAG_TRACK_PEAK, track_peak, GST_TAG_TRACK_GAIN, track_gain, NULL); - } - - return track_success; -} - -static gboolean -gst_rg_analysis_album_result (GstRgAnalysis * filter, GstTagList ** tag_list) -{ - gboolean album_success; - gdouble album_gain, album_peak; - - album_success = rg_analysis_album_result (filter->ctx, &album_gain, - &album_peak); - - if (album_success) { - album_gain += filter->reference_level - RG_REFERENCE_LEVEL; - GST_INFO_OBJECT (filter, "album gain is %+.2f dB, peak %.6f", album_gain, - album_peak); - } else { - GST_INFO_OBJECT (filter, "album was too short to analyze"); - } - - if (album_success) { - if (*tag_list == NULL) - *tag_list = gst_tag_list_new (); - gst_tag_list_add (*tag_list, GST_TAG_MERGE_APPEND, - GST_TAG_ALBUM_PEAK, album_peak, GST_TAG_ALBUM_GAIN, album_gain, NULL); - } - - return album_success; -} diff --git a/gst/replaygain/gstrganalysis.h b/gst/replaygain/gstrganalysis.h deleted file mode 100644 index fbf46830..00000000 --- a/gst/replaygain/gstrganalysis.h +++ /dev/null @@ -1,85 +0,0 @@ -/* GStreamer ReplayGain analysis - * - * Copyright (C) 2006 Rene Stadler <mail@renestadler.de> - * - * gstrganalysis.h: Element that performs the ReplayGain analysis - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifndef __GST_RG_ANALYSIS_H__ -#define __GST_RG_ANALYSIS_H__ - -#include <gst/gst.h> -#include <gst/base/gstbasetransform.h> - -#include "rganalysis.h" - -G_BEGIN_DECLS - -#define GST_TYPE_RG_ANALYSIS \ - (gst_rg_analysis_get_type()) -#define GST_RG_ANALYSIS(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_ANALYSIS,GstRgAnalysis)) -#define GST_RG_ANALYSIS_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_ANALYSIS,GstRgAnalysisClass)) -#define GST_IS_RG_ANALYSIS(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_ANALYSIS)) -#define GST_IS_RG_ANALYSIS_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_ANALYSIS)) -typedef struct _GstRgAnalysis GstRgAnalysis; -typedef struct _GstRgAnalysisClass GstRgAnalysisClass; - -/** - * GstRgAnalysis: - * - * Opaque data structure. - */ -struct _GstRgAnalysis -{ - GstBaseTransform element; - - /*< private >*/ - - RgAnalysisCtx *ctx; - void (*analyze) (RgAnalysisCtx * ctx, gconstpointer data, gsize size, - guint depth); - gint depth; - - /* Property values. */ - guint num_tracks; - gdouble reference_level; - gboolean forced; - - /* State machinery for skipping. */ - gboolean ignore_tags; - gboolean skip; - gboolean has_track_gain; - gboolean has_track_peak; - gboolean has_album_gain; - gboolean has_album_peak; -}; - -struct _GstRgAnalysisClass -{ - GstBaseTransformClass parent_class; -}; - -GType gst_rg_analysis_get_type (void); - -G_END_DECLS - -#endif /* __GST_RG_ANALYSIS_H__ */ diff --git a/gst/replaygain/gstrglimiter.c b/gst/replaygain/gstrglimiter.c deleted file mode 100644 index 43c7b01a..00000000 --- a/gst/replaygain/gstrglimiter.c +++ /dev/null @@ -1,202 +0,0 @@ -/* GStreamer ReplayGain limiter - * - * Copyright (C) 2007 Rene Stadler <mail@renestadler.de> - * - * gstrglimiter.c: Element to apply signal compression to raw audio data - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -/** - * SECTION:element-rglimiter - * @see_also: #GstRgVolume - * - * This element applies signal compression/limiting to raw audio data. It - * performs strict hard limiting with soft-knee characteristics, using a - * threshold of -6 dB. This type of filter is mentioned in the proposed <ulink - * url="http://replaygain.org">ReplayGain standard</ulink>. - * - * <refsect2> - * <title>Example launch line</title> - * |[ - * gst-launch filesrc location=filename.ext ! decodebin ! audioconvert \ - * ! rgvolume pre-amp=6.0 headroom=10.0 ! rglimiter \ - * ! audioconvert ! audioresample ! alsasink - * ]|Playback of a file - * </refsect2> - */ - -#ifdef HAVE_CONFIG_H -#include <config.h> -#endif - -#include <gst/gst.h> -#include <math.h> - -#include "gstrglimiter.h" - -GST_DEBUG_CATEGORY_STATIC (gst_rg_limiter_debug); -#define GST_CAT_DEFAULT gst_rg_limiter_debug - -enum -{ - PROP_0, - PROP_ENABLED, -}; - -static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "width = (int) 32, channels = (int) [1, MAX], " - "rate = (int) [1, MAX], endianness = (int) BYTE_ORDER")); - -static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "width = (int) 32, channels = (int) [1, MAX], " - "rate = (int) [1, MAX], endianness = (int) BYTE_ORDER")); - -GST_BOILERPLATE (GstRgLimiter, gst_rg_limiter, GstBaseTransform, - GST_TYPE_BASE_TRANSFORM); - -static void gst_rg_limiter_class_init (GstRgLimiterClass * klass); -static void gst_rg_limiter_init (GstRgLimiter * filter, - GstRgLimiterClass * gclass); - -static void gst_rg_limiter_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_rg_limiter_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); - -static GstFlowReturn gst_rg_limiter_transform_ip (GstBaseTransform * base, - GstBuffer * buf); - -static const GstElementDetails element_details = { - "ReplayGain limiter", - "Filter/Effect/Audio", - "Apply signal compression to raw audio data", - "Ren\xc3\xa9 Stadler <mail@renestadler.de>" -}; - -static void -gst_rg_limiter_base_init (gpointer g_class) -{ - GstElementClass *element_class = g_class; - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_factory)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_factory)); - gst_element_class_set_details (element_class, &element_details); - - GST_DEBUG_CATEGORY_INIT (gst_rg_limiter_debug, "rglimiter", 0, - "ReplayGain limiter element"); -} - -static void -gst_rg_limiter_class_init (GstRgLimiterClass * klass) -{ - GObjectClass *gobject_class; - GstBaseTransformClass *trans_class; - - gobject_class = (GObjectClass *) klass; - - gobject_class->set_property = gst_rg_limiter_set_property; - gobject_class->get_property = gst_rg_limiter_get_property; - - g_object_class_install_property (gobject_class, PROP_ENABLED, - g_param_spec_boolean ("enabled", "Enabled", "Enable processing", TRUE, - G_PARAM_READWRITE)); - - trans_class = GST_BASE_TRANSFORM_CLASS (klass); - trans_class->transform_ip = GST_DEBUG_FUNCPTR (gst_rg_limiter_transform_ip); - trans_class->passthrough_on_same_caps = FALSE; -} - -static void -gst_rg_limiter_init (GstRgLimiter * filter, GstRgLimiterClass * gclass) -{ - GstBaseTransform *base = GST_BASE_TRANSFORM (filter); - - gst_base_transform_set_passthrough (base, FALSE); - gst_base_transform_set_gap_aware (base, TRUE); - - filter->enabled = TRUE; -} - -static void -gst_rg_limiter_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstRgLimiter *filter = GST_RG_LIMITER (object); - - switch (prop_id) { - case PROP_ENABLED: - filter->enabled = g_value_get_boolean (value); - gst_base_transform_set_passthrough (GST_BASE_TRANSFORM (filter), - !filter->enabled); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_rg_limiter_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstRgLimiter *filter = GST_RG_LIMITER (object); - - switch (prop_id) { - case PROP_ENABLED: - g_value_set_boolean (value, filter->enabled); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -#define LIMIT 1.0 -#define THRES 0.5 /* ca. -6 dB */ -#define COMPL 0.5 /* LIMIT - THRESH */ - -static GstFlowReturn -gst_rg_limiter_transform_ip (GstBaseTransform * base, GstBuffer * buf) -{ - GstRgLimiter *filter = GST_RG_LIMITER (base); - gfloat *input; - guint count; - guint i; - - if (!filter->enabled) - return GST_FLOW_OK; - - if (GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_GAP)) - return GST_FLOW_OK; - - input = (gfloat *) GST_BUFFER_DATA (buf); - count = GST_BUFFER_SIZE (buf) / sizeof (gfloat); - - for (i = count; i--;) { - if (*input > THRES) - *input = tanhf ((*input - THRES) / COMPL) * COMPL + THRES; - else if (*input < -THRES) - *input = tanhf ((*input + THRES) / COMPL) * COMPL - THRES; - input++; - } - - return GST_FLOW_OK; -} diff --git a/gst/replaygain/gstrglimiter.h b/gst/replaygain/gstrglimiter.h deleted file mode 100644 index 63bd8049..00000000 --- a/gst/replaygain/gstrglimiter.h +++ /dev/null @@ -1,64 +0,0 @@ -/* GStreamer ReplayGain limiter - * - * Copyright (C) 2007 Rene Stadler <mail@renestadler.de> - * - * gstrglimiter.h: Element to apply signal compression to raw audio data - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifndef __GST_RG_LIMITER_H__ -#define __GST_RG_LIMITER_H__ - -#include <gst/gst.h> -#include <gst/base/gstbasetransform.h> - -#define GST_TYPE_RG_LIMITER \ - (gst_rg_limiter_get_type()) -#define GST_RG_LIMITER(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_LIMITER,GstRgLimiter)) -#define GST_RG_LIMITER_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_LIMITER,GstRgLimiterClass)) -#define GST_IS_RG_LIMITER(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_LIMITER)) -#define GST_IS_RG_LIMITER_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_LIMITER)) - -typedef struct _GstRgLimiter GstRgLimiter; -typedef struct _GstRgLimiterClass GstRgLimiterClass; - -/** - * GstRgLimiter: - * - * Opaque data structure. - */ -struct _GstRgLimiter -{ - GstBaseTransform element; - - /*< private >*/ - - gboolean enabled; -}; - -struct _GstRgLimiterClass -{ - GstBaseTransformClass parent_class; -}; - -GType gst_rg_limiter_get_type (void); - -#endif /* __GST_RG_LIMITER_H__ */ diff --git a/gst/replaygain/gstrgvolume.c b/gst/replaygain/gstrgvolume.c deleted file mode 100644 index 41fe441d..00000000 --- a/gst/replaygain/gstrgvolume.c +++ /dev/null @@ -1,698 +0,0 @@ -/* GStreamer ReplayGain volume adjustment - * - * Copyright (C) 2007 Rene Stadler <mail@renestadler.de> - * - * gstrgvolume.c: Element to apply ReplayGain volume adjustment - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -/** - * SECTION:element-rgvolume - * @see_also: #GstRgLimiter, #GstRgAnalysis - * - * This element applies volume changes to streams as lined out in the proposed - * <ulink url="http://replaygain.org">ReplayGain standard</ulink>. It - * interprets the ReplayGain meta data tags and carries out the adjustment (by - * using a volume element internally). The relevant tags are: - * <itemizedlist> - * <listitem>#GST_TAG_TRACK_GAIN</listitem> - * <listitem>#GST_TAG_TRACK_PEAK</listitem> - * <listitem>#GST_TAG_ALBUM_GAIN</listitem> - * <listitem>#GST_TAG_ALBUM_PEAK</listitem> - * <listitem>#GST_TAG_REFERENCE_LEVEL</listitem> - * </itemizedlist> - * The information carried by these tags must have been calculated beforehand by - * performing the ReplayGain analysis. This is implemented by the <link - * linkend="GstRgAnalysis">rganalysis</link> element. - * - * The signal compression/limiting recommendations outlined in the proposed - * standard are not implemented by this element. This has to be handled by - * separate elements because applications might want to have additional filters - * between the volume adjustment and the limiting stage. A basic limiter is - * included with this plugin: The <link linkend="GstRgLimiter">rglimiter</link> - * element applies -6 dB hard limiting as mentioned in the ReplayGain standard. - * - * <refsect2> - * <title>Example launch line</title> - * |[ - * gst-launch filesrc location=filename.ext ! decodebin ! audioconvert \ - * ! rgvolume ! audioconvert ! audioresample ! alsasink - * ]| Playback of a file - * </refsect2> - */ - -#ifdef HAVE_CONFIG_H -#include <config.h> -#endif - -#include <gst/gst.h> -#include <gst/pbutils/pbutils.h> -#include <math.h> - -#include "gstrgvolume.h" -#include "replaygain.h" - -GST_DEBUG_CATEGORY_STATIC (gst_rg_volume_debug); -#define GST_CAT_DEFAULT gst_rg_volume_debug - -enum -{ - PROP_0, - PROP_ALBUM_MODE, - PROP_HEADROOM, - PROP_PRE_AMP, - PROP_FALLBACK_GAIN, - PROP_TARGET_GAIN, - PROP_RESULT_GAIN -}; - -#define DEFAULT_ALBUM_MODE TRUE -#define DEFAULT_HEADROOM 0.0 -#define DEFAULT_PRE_AMP 0.0 -#define DEFAULT_FALLBACK_GAIN 0.0 - -#define DB_TO_LINEAR(x) pow (10., (x) / 20.) -#define LINEAR_TO_DB(x) (20. * log10 (x)) - -#define GAIN_FORMAT "+.02f dB" -#define PEAK_FORMAT ".06f" - -#define VALID_GAIN(x) ((x) > -60.00 && (x) < 60.00) -#define VALID_PEAK(x) ((x) > 0.) - -/* Same template caps as GstVolume, for I don't like having just ANY caps. */ - -static GstStaticPadTemplate sink_template = GST_STATIC_PAD_TEMPLATE ("sink", - GST_PAD_SINK, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, MAX ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) 32; " - "audio/x-raw-int, " - "channels = (int) [ 1, MAX ], " - "rate = (int) [ 1, MAX ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE")); - -static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src", - GST_PAD_SRC, GST_PAD_ALWAYS, GST_STATIC_CAPS ("audio/x-raw-float, " - "rate = (int) [ 1, MAX ], " - "channels = (int) [ 1, MAX ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) 32; " - "audio/x-raw-int, " - "channels = (int) [ 1, MAX ], " - "rate = (int) [ 1, MAX ], " - "endianness = (int) BYTE_ORDER, " - "width = (int) 16, " "depth = (int) 16, " "signed = (bool) TRUE")); - -GST_BOILERPLATE (GstRgVolume, gst_rg_volume, GstBin, GST_TYPE_BIN); - -static void gst_rg_volume_class_init (GstRgVolumeClass * klass); -static void gst_rg_volume_init (GstRgVolume * self, GstRgVolumeClass * gclass); - -static void gst_rg_volume_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec); -static void gst_rg_volume_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec); -static void gst_rg_volume_dispose (GObject * object); - -static GstStateChangeReturn gst_rg_volume_change_state (GstElement * element, - GstStateChange transition); -static gboolean gst_rg_volume_sink_event (GstPad * pad, GstEvent * event); - -static GstEvent *gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event); -static void gst_rg_volume_reset (GstRgVolume * self); -static void gst_rg_volume_update_gain (GstRgVolume * self); -static inline void gst_rg_volume_determine_gain (GstRgVolume * self, - gdouble * target_gain, gdouble * result_gain); - -static void -gst_rg_volume_base_init (gpointer g_class) -{ - GstElementClass *element_class = g_class; - - static const GstElementDetails element_details = { - "ReplayGain volume", - "Filter/Effect/Audio", - "Apply ReplayGain volume adjustment", - "Ren\xc3\xa9 Stadler <mail@renestadler.de>" - }; - - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&src_template)); - gst_element_class_add_pad_template (element_class, - gst_static_pad_template_get (&sink_template)); - gst_element_class_set_details (element_class, &element_details); - - GST_DEBUG_CATEGORY_INIT (gst_rg_volume_debug, "rgvolume", 0, - "ReplayGain volume element"); -} - -static void -gst_rg_volume_class_init (GstRgVolumeClass * klass) -{ - GObjectClass *gobject_class; - GstElementClass *element_class; - GstBinClass *bin_class; - - gobject_class = (GObjectClass *) klass; - - gobject_class->set_property = gst_rg_volume_set_property; - gobject_class->get_property = gst_rg_volume_get_property; - gobject_class->dispose = gst_rg_volume_dispose; - - /** - * GstRgVolume:album-mode: - * - * Whether to prefer album gain over track gain. - * - * If set to %TRUE, use album gain instead of track gain if both are - * available. This keeps the relative loudness levels of tracks from the same - * album intact. - * - * If set to %FALSE, track mode is used instead. This effectively leads to - * more extensive normalization. - * - * If album mode is enabled but the album gain tag is absent in the stream, - * the track gain is used instead. If both gain tags are missing, the value - * of the <link linkend="GstRgVolume--fallback-gain">fallback-gain</link> - * property is used instead. - */ - g_object_class_install_property (gobject_class, PROP_ALBUM_MODE, - g_param_spec_boolean ("album-mode", "Album mode", - "Prefer album over track gain", DEFAULT_ALBUM_MODE, - G_PARAM_READWRITE)); - /** - * GstRgVolume:headroom: - * - * Extra headroom [dB]. This controls the amount by which the output can - * exceed digital full scale. - * - * Only set this to a value greater than 0.0 if signal compression/limiting of - * a suitable form is applied to the output (or output is brought into the - * correct range by some other transformation). - * - * This element internally uses a volume element, which also supports - * operating on integer audio formats. These formats do not allow exceeding - * digital full scale. If extra headroom is used, make sure that the raw - * audio data format is floating point (audio/x-raw-float). Otherwise, - * clipping distortion might be introduced as part of the volume adjustment - * itself. - */ - g_object_class_install_property (gobject_class, PROP_HEADROOM, - g_param_spec_double ("headroom", "Headroom", "Extra headroom [dB]", - 0., 60., DEFAULT_HEADROOM, G_PARAM_READWRITE)); - /** - * GstRgVolume:pre-amp: - * - * Additional gain to apply globally [dB]. This controls the trade-off - * between uniformity of normalization and utilization of available dynamic - * range. - * - * Note that the default value is 0 dB because the ReplayGain reference value - * was adjusted by +6 dB (from 83 to 89 dB). At the time of this writing, the - * <ulink url="http://replaygain.org">webpage</ulink> is still outdated and - * does not reflect this change however. Where the original proposal states - * that a proper default pre-amp value is +6 dB, this translates to the used 0 - * dB. - */ - g_object_class_install_property (gobject_class, PROP_PRE_AMP, - g_param_spec_double ("pre-amp", "Pre-amp", "Extra gain [dB]", - -60., 60., DEFAULT_PRE_AMP, G_PARAM_READWRITE)); - /** - * GstRgVolume:fallback-gain: - * - * Fallback gain [dB] for streams missing ReplayGain tags. - */ - g_object_class_install_property (gobject_class, PROP_FALLBACK_GAIN, - g_param_spec_double ("fallback-gain", "Fallback gain", - "Gain for streams missing tags [dB]", - -60., 60., DEFAULT_FALLBACK_GAIN, G_PARAM_READWRITE)); - /** - * GstRgVolume:result-gain: - * - * Applied gain [dB]. This gain is applied to processed buffer data. - * - * This is set to the <link linkend="GstRgVolume--target-gain">target - * gain</link> if amplification by that amount can be applied safely. - * "Safely" means that the volume adjustment does not inflict clipping - * distortion. Should this not be the case, the result gain is set to an - * appropriately reduced value (by applying peak normalization). The proposed - * standard calls this "clipping prevention". - * - * The difference between target and result gain reflects the necessary amount - * of reduction. Applications can make use of this information to temporarily - * reduce the <link linkend="GstRgVolume--pre-amp">pre-amp</link> for - * subsequent streams, as recommended by the ReplayGain standard. - * - * Note that target and result gain differing for a great majority of streams - * indicates a problem: What happens in this case is that most streams receive - * peak normalization instead of amplification by the ideal replay gain. To - * prevent this, the <link linkend="GstRgVolume--pre-amp">pre-amp</link> has - * to be lowered and/or a limiter has to be used which facilitates the use of - * <link linkend="GstRgVolume--headroom">headroom</link>. - */ - g_object_class_install_property (gobject_class, PROP_RESULT_GAIN, - g_param_spec_double ("result-gain", "Result-gain", "Applied gain [dB]", - -120., 120., 0., G_PARAM_READABLE)); - /** - * GstRgVolume:target-gain: - * - * Applicable gain [dB]. This gain is supposed to be applied. - * - * Depending on the value of the <link - * linkend="GstRgVolume--album-mode">album-mode</link> property and the - * presence of ReplayGain tags in the stream, this is set according to one of - * these simple formulas: - * - * <itemizedlist> - * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + album gain - * of the stream</listitem> - * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + track gain - * of the stream</listitem> - * <listitem><link linkend="GstRgVolume--pre-amp">pre-amp</link> + <link - * linkend="GstRgVolume--fallback-gain">fallback gain</link></listitem> - * </itemizedlist> - */ - g_object_class_install_property (gobject_class, PROP_TARGET_GAIN, - g_param_spec_double ("target-gain", "Target-gain", - "Applicable gain [dB]", -120., 120., 0., G_PARAM_READABLE)); - - element_class = (GstElementClass *) klass; - element_class->change_state = GST_DEBUG_FUNCPTR (gst_rg_volume_change_state); - - bin_class = (GstBinClass *) klass; - /* Setting these to NULL makes gst_bin_add and _remove refuse to let anyone - * mess with our internals. */ - bin_class->add_element = NULL; - bin_class->remove_element = NULL; -} - -static void -gst_rg_volume_init (GstRgVolume * self, GstRgVolumeClass * gclass) -{ - GObjectClass *volume_class; - GstPad *volume_pad, *ghost_pad; - - self->album_mode = DEFAULT_ALBUM_MODE; - self->headroom = DEFAULT_HEADROOM; - self->pre_amp = DEFAULT_PRE_AMP; - self->fallback_gain = DEFAULT_FALLBACK_GAIN; - self->target_gain = 0.0; - self->result_gain = 0.0; - - self->volume_element = gst_element_factory_make ("volume", "rgvolume-volume"); - if (G_UNLIKELY (self->volume_element == NULL)) { - GstMessage *msg; - - GST_WARNING_OBJECT (self, "could not create volume element"); - msg = gst_missing_element_message_new (GST_ELEMENT_CAST (self), "volume"); - gst_element_post_message (GST_ELEMENT_CAST (self), msg); - - /* Nothing else to do, we will refuse the state change from NULL to READY to - * indicate that something went very wrong. It is doubtful that someone - * attempts changing our state though, since we end up having no pads! */ - return; - } - - volume_class = G_OBJECT_GET_CLASS (G_OBJECT (self->volume_element)); - self->max_volume = G_PARAM_SPEC_DOUBLE - (g_object_class_find_property (volume_class, "volume"))->maximum; - - GST_BIN_CLASS (parent_class)->add_element (GST_BIN_CAST (self), - self->volume_element); - - volume_pad = gst_element_get_static_pad (self->volume_element, "sink"); - ghost_pad = gst_ghost_pad_new_from_template ("sink", volume_pad, - gst_pad_get_pad_template (volume_pad)); - gst_object_unref (volume_pad); - gst_pad_set_event_function (ghost_pad, gst_rg_volume_sink_event); - gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad); - - volume_pad = gst_element_get_static_pad (self->volume_element, "src"); - ghost_pad = gst_ghost_pad_new_from_template ("src", volume_pad, - gst_pad_get_pad_template (volume_pad)); - gst_object_unref (volume_pad); - gst_element_add_pad (GST_ELEMENT_CAST (self), ghost_pad); -} - -static void -gst_rg_volume_set_property (GObject * object, guint prop_id, - const GValue * value, GParamSpec * pspec) -{ - GstRgVolume *self = GST_RG_VOLUME (object); - - switch (prop_id) { - case PROP_ALBUM_MODE: - self->album_mode = g_value_get_boolean (value); - break; - case PROP_HEADROOM: - self->headroom = g_value_get_double (value); - break; - case PROP_PRE_AMP: - self->pre_amp = g_value_get_double (value); - break; - case PROP_FALLBACK_GAIN: - self->fallback_gain = g_value_get_double (value); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } - - gst_rg_volume_update_gain (self); -} - -static void -gst_rg_volume_get_property (GObject * object, guint prop_id, - GValue * value, GParamSpec * pspec) -{ - GstRgVolume *self = GST_RG_VOLUME (object); - - switch (prop_id) { - case PROP_ALBUM_MODE: - g_value_set_boolean (value, self->album_mode); - break; - case PROP_HEADROOM: - g_value_set_double (value, self->headroom); - break; - case PROP_PRE_AMP: - g_value_set_double (value, self->pre_amp); - break; - case PROP_FALLBACK_GAIN: - g_value_set_double (value, self->fallback_gain); - break; - case PROP_TARGET_GAIN: - g_value_set_double (value, self->target_gain); - break; - case PROP_RESULT_GAIN: - g_value_set_double (value, self->result_gain); - break; - default: - G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); - break; - } -} - -static void -gst_rg_volume_dispose (GObject * object) -{ - GstRgVolume *self = GST_RG_VOLUME (object); - - if (self->volume_element != NULL) { - /* Manually remove our child using the bin implementation of remove_element. - * This is needed because we prevent gst_bin_remove from working, which the - * parent dispose handler would use if we had any children left. */ - GST_BIN_CLASS (parent_class)->remove_element (GST_BIN_CAST (self), - self->volume_element); - self->volume_element = NULL; - } - - G_OBJECT_CLASS (parent_class)->dispose (object); -} - -static GstStateChangeReturn -gst_rg_volume_change_state (GstElement * element, GstStateChange transition) -{ - GstRgVolume *self = GST_RG_VOLUME (element); - GstStateChangeReturn res; - - switch (transition) { - case GST_STATE_CHANGE_NULL_TO_READY: - - if (G_UNLIKELY (self->volume_element == NULL)) { - /* Creating our child volume element in _init failed. */ - return GST_STATE_CHANGE_FAILURE; - } - break; - - case GST_STATE_CHANGE_READY_TO_PAUSED: - - gst_rg_volume_reset (self); - break; - - default: - break; - } - - res = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition); - - return res; -} - -/* Event function for the ghost sink pad. */ -static gboolean -gst_rg_volume_sink_event (GstPad * pad, GstEvent * event) -{ - GstRgVolume *self; - GstPad *volume_sink_pad; - GstEvent *send_event = event; - gboolean res; - - self = GST_RG_VOLUME (gst_pad_get_parent_element (pad)); - volume_sink_pad = gst_ghost_pad_get_target (GST_GHOST_PAD (pad)); - - switch (GST_EVENT_TYPE (event)) { - case GST_EVENT_TAG: - - GST_LOG_OBJECT (self, "received tag event"); - - send_event = gst_rg_volume_tag_event (self, event); - - if (send_event == NULL) - GST_LOG_OBJECT (self, "all tags handled, dropping event"); - - break; - - case GST_EVENT_EOS: - - gst_rg_volume_reset (self); - break; - - default: - break; - } - - if (G_LIKELY (send_event != NULL)) - res = gst_pad_send_event (volume_sink_pad, send_event); - else - res = TRUE; - - gst_object_unref (volume_sink_pad); - gst_object_unref (self); - return res; -} - -static GstEvent * -gst_rg_volume_tag_event (GstRgVolume * self, GstEvent * event) -{ - GstTagList *tag_list; - gboolean has_track_gain, has_track_peak, has_album_gain, has_album_peak; - gboolean has_ref_level; - - g_return_val_if_fail (event != NULL, NULL); - g_return_val_if_fail (GST_EVENT_TYPE (event) == GST_EVENT_TAG, event); - - gst_event_parse_tag (event, &tag_list); - - if (gst_tag_list_is_empty (tag_list)) - return event; - - has_track_gain = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_GAIN, - &self->track_gain); - has_track_peak = gst_tag_list_get_double (tag_list, GST_TAG_TRACK_PEAK, - &self->track_peak); - has_album_gain = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_GAIN, - &self->album_gain); - has_album_peak = gst_tag_list_get_double (tag_list, GST_TAG_ALBUM_PEAK, - &self->album_peak); - has_ref_level = gst_tag_list_get_double (tag_list, GST_TAG_REFERENCE_LEVEL, - &self->reference_level); - - if (!has_track_gain && !has_track_peak && !has_album_gain && !has_album_peak) - return event; - - if (has_ref_level && (has_track_gain || has_album_gain) - && (ABS (self->reference_level - RG_REFERENCE_LEVEL) > 1.e-6)) { - /* Log a message stating the amount of adjustment that is applied below. */ - GST_DEBUG_OBJECT (self, - "compensating for reference level difference by %" GAIN_FORMAT, - RG_REFERENCE_LEVEL - self->reference_level); - } - if (has_track_gain) { - self->track_gain += RG_REFERENCE_LEVEL - self->reference_level; - } - if (has_album_gain) { - self->album_gain += RG_REFERENCE_LEVEL - self->reference_level; - } - - /* Ignore values that are obviously invalid. */ - if (G_UNLIKELY (has_track_gain && !VALID_GAIN (self->track_gain))) { - GST_DEBUG_OBJECT (self, - "ignoring bogus track gain value %" GAIN_FORMAT, self->track_gain); - has_track_gain = FALSE; - } - if (G_UNLIKELY (has_track_peak && !VALID_PEAK (self->track_peak))) { - GST_DEBUG_OBJECT (self, - "ignoring bogus track peak value %" PEAK_FORMAT, self->track_peak); - has_track_peak = FALSE; - } - if (G_UNLIKELY (has_album_gain && !VALID_GAIN (self->album_gain))) { - GST_DEBUG_OBJECT (self, - "ignoring bogus album gain value %" GAIN_FORMAT, self->album_gain); - has_album_gain = FALSE; - } - if (G_UNLIKELY (has_album_peak && !VALID_PEAK (self->album_peak))) { - GST_DEBUG_OBJECT (self, - "ignoring bogus album peak value %" PEAK_FORMAT, self->album_peak); - has_album_peak = FALSE; - } - - self->has_track_gain |= has_track_gain; - self->has_track_peak |= has_track_peak; - self->has_album_gain |= has_album_gain; - self->has_album_peak |= has_album_peak; - - event = (GstEvent *) gst_mini_object_make_writable (GST_MINI_OBJECT (event)); - gst_event_parse_tag (event, &tag_list); - - gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_GAIN); - gst_tag_list_remove_tag (tag_list, GST_TAG_TRACK_PEAK); - gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_GAIN); - gst_tag_list_remove_tag (tag_list, GST_TAG_ALBUM_PEAK); - gst_tag_list_remove_tag (tag_list, GST_TAG_REFERENCE_LEVEL); - - gst_rg_volume_update_gain (self); - - if (gst_tag_list_is_empty (tag_list)) { - gst_event_unref (event); - event = NULL; - } - - return event; -} - -static void -gst_rg_volume_reset (GstRgVolume * self) -{ - self->has_track_gain = FALSE; - self->has_track_peak = FALSE; - self->has_album_gain = FALSE; - self->has_album_peak = FALSE; - - self->reference_level = RG_REFERENCE_LEVEL; - - gst_rg_volume_update_gain (self); -} - -static void -gst_rg_volume_update_gain (GstRgVolume * self) -{ - gdouble target_gain, result_gain, result_volume; - gboolean target_gain_changed, result_gain_changed; - - gst_rg_volume_determine_gain (self, &target_gain, &result_gain); - - result_volume = DB_TO_LINEAR (result_gain); - - /* Ensure that the result volume is within the range that the volume element - * can handle. Currently, the limit is 10. (+20 dB), which should not be - * restrictive. */ - if (G_UNLIKELY (result_volume > self->max_volume)) { - GST_INFO_OBJECT (self, - "cannot handle result gain of %" GAIN_FORMAT " (%0.6f), adjusting", - result_gain, result_volume); - - result_volume = self->max_volume; - result_gain = LINEAR_TO_DB (result_volume); - } - - /* Direct comparison is OK in this case. */ - if (target_gain == result_gain) { - GST_INFO_OBJECT (self, - "result gain is %" GAIN_FORMAT " (%0.6f), matching target", - result_gain, result_volume); - } else { - GST_INFO_OBJECT (self, - "result gain is %" GAIN_FORMAT " (%0.6f), target is %" GAIN_FORMAT, - result_gain, result_volume, target_gain); - } - - target_gain_changed = (self->target_gain != target_gain); - result_gain_changed = (self->result_gain != result_gain); - - self->target_gain = target_gain; - self->result_gain = result_gain; - - g_object_set (self->volume_element, "volume", result_volume, NULL); - - if (target_gain_changed) - g_object_notify ((GObject *) self, "target-gain"); - if (result_gain_changed) - g_object_notify ((GObject *) self, "result-gain"); -} - -static inline void -gst_rg_volume_determine_gain (GstRgVolume * self, gdouble * target_gain, - gdouble * result_gain) -{ - gdouble gain, peak; - - if (!self->has_track_gain && !self->has_album_gain) { - - GST_DEBUG_OBJECT (self, "using fallback gain"); - gain = self->fallback_gain; - peak = 1.0; - - } else if ((self->album_mode && self->has_album_gain) - || (!self->album_mode && !self->has_track_gain)) { - - gain = self->album_gain; - if (G_LIKELY (self->has_album_peak)) { - peak = self->album_peak; - } else { - GST_DEBUG_OBJECT (self, "album peak missing, assuming 1.0"); - peak = 1.0; - } - /* Falling back from track to album gain shouldn't really happen. */ - if (G_UNLIKELY (!self->album_mode)) - GST_INFO_OBJECT (self, "falling back to album gain"); - - } else { - /* !album_mode && !has_album_gain || album_mode && has_track_gain */ - - gain = self->track_gain; - if (G_LIKELY (self->has_track_peak)) { - peak = self->track_peak; - } else { - GST_DEBUG_OBJECT (self, "track peak missing, assuming 1.0"); - peak = 1.0; - } - if (self->album_mode) - GST_INFO_OBJECT (self, "falling back to track gain"); - } - - gain += self->pre_amp; - - *target_gain = gain; - *result_gain = gain; - - if (LINEAR_TO_DB (peak) + gain > self->headroom) { - *result_gain = LINEAR_TO_DB (1. / peak) + self->headroom; - } -} diff --git a/gst/replaygain/gstrgvolume.h b/gst/replaygain/gstrgvolume.h deleted file mode 100644 index a0a5a8ce..00000000 --- a/gst/replaygain/gstrgvolume.h +++ /dev/null @@ -1,88 +0,0 @@ -/* GStreamer ReplayGain volume adjustment - * - * Copyright (C) 2007 Rene Stadler <mail@renestadler.de> - * - * gstrgvolume.h: Element to apply ReplayGain volume adjustment - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifndef __GST_RG_VOLUME_H__ -#define __GST_RG_VOLUME_H__ - -#include <gst/gst.h> - -G_BEGIN_DECLS - -#define GST_TYPE_RG_VOLUME \ - (gst_rg_volume_get_type()) -#define GST_RG_VOLUME(obj) \ - (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_RG_VOLUME,GstRgVolume)) -#define GST_RG_VOLUME_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_RG_VOLUME,GstRgVolumeClass)) -#define GST_IS_RG_VOLUME(obj) \ - (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_RG_VOLUME)) -#define GST_IS_RG_VOLUME_CLASS(klass) \ - (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_RG_VOLUME)) - -typedef struct _GstRgVolume GstRgVolume; -typedef struct _GstRgVolumeClass GstRgVolumeClass; - -/** - * GstRgVolume: - * - * Opaque data structure. - */ -struct _GstRgVolume -{ - GstBin bin; - - /*< private >*/ - - GstElement *volume_element; - gdouble max_volume; - - gboolean album_mode; - gdouble headroom; - gdouble pre_amp; - gdouble fallback_gain; - - gdouble target_gain; - gdouble result_gain; - - gdouble track_gain; - gdouble track_peak; - gdouble album_gain; - gdouble album_peak; - - gboolean has_track_gain; - gboolean has_track_peak; - gboolean has_album_gain; - gboolean has_album_peak; - - gdouble reference_level; -}; - -struct _GstRgVolumeClass -{ - GstBinClass parent_class; -}; - -GType gst_rg_volume_get_type (void); - -G_END_DECLS - -#endif /* __GST_RG_VOLUME_H__ */ diff --git a/gst/replaygain/replaygain.c b/gst/replaygain/replaygain.c deleted file mode 100644 index d0127e8b..00000000 --- a/gst/replaygain/replaygain.c +++ /dev/null @@ -1,53 +0,0 @@ -/* GStreamer ReplayGain plugin - * - * Copyright (C) 2006 Rene Stadler <mail@renestadler.de> - * - * replaygain.c: Plugin providing ReplayGain related elements - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifdef HAVE_CONFIG_H -#include <config.h> -#endif - -#include <gst/gst.h> - -#include "gstrganalysis.h" -#include "gstrglimiter.h" -#include "gstrgvolume.h" - -static gboolean -plugin_init (GstPlugin * plugin) -{ - if (!gst_element_register (plugin, "rganalysis", GST_RANK_NONE, - GST_TYPE_RG_ANALYSIS)) - return FALSE; - - if (!gst_element_register (plugin, "rglimiter", GST_RANK_NONE, - GST_TYPE_RG_LIMITER)) - return FALSE; - - if (!gst_element_register (plugin, "rgvolume", GST_RANK_NONE, - GST_TYPE_RG_VOLUME)) - return FALSE; - - return TRUE; -} - -GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, GST_VERSION_MINOR, "replaygain", - "ReplayGain volume normalization", plugin_init, VERSION, GST_LICENSE, - GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN); diff --git a/gst/replaygain/replaygain.h b/gst/replaygain/replaygain.h deleted file mode 100644 index 15be8885..00000000 --- a/gst/replaygain/replaygain.h +++ /dev/null @@ -1,36 +0,0 @@ -/* GStreamer ReplayGain plugin - * - * Copyright (C) 2006 Rene Stadler <mail@renestadler.de> - * - * replaygain.h: Plugin providing ReplayGain related elements - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifndef __REPLAYGAIN_H__ -#define __REPLAYGAIN_H__ - -G_BEGIN_DECLS - -/* Reference level (in dBSPL). The 2001 proposal specifies 83. This was - * changed later in all implementations to 89, which is the new, offical value: - * David Robinson acknowledged the change but didn't update the website yet. */ - -#define RG_REFERENCE_LEVEL 89. - -G_END_DECLS - -#endif /* __REPLAYGAIN_H__ */ diff --git a/gst/replaygain/rganalysis.c b/gst/replaygain/rganalysis.c deleted file mode 100644 index 147eef85..00000000 --- a/gst/replaygain/rganalysis.c +++ /dev/null @@ -1,777 +0,0 @@ -/* GStreamer ReplayGain analysis - * - * Copyright (C) 2006 Rene Stadler <mail@renestadler.de> - * Copyright (C) 2001 David Robinson <David@Robinson.org> - * Glen Sawyer <glensawyer@hotmail.com> - * - * rganalysis.c: Analyze raw audio data in accordance with ReplayGain - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -/* Based on code with Copyright (C) 2001 David Robinson - * <David@Robinson.org> and Glen Sawyer <glensawyer@hotmail.com>, - * which is distributed under the LGPL as part of the vorbisgain - * program. The original code also mentions Frank Klemm - * (http://www.uni-jena.de/~pfk/mpp/) for having contributed lots of - * good code. Specifically, this is based on the file - * "gain_analysis.c" from vorbisgain version 0.34. - */ - -/* Room for future improvement: Mono data is currently in fact copied - * to two channels which get processed normally. This means that mono - * input data is processed twice. - */ - -/* Helpful information for understanding this code: The two IIR - * filters depend on previous input _and_ previous output samples (up - * to the filter's order number of samples). This explains the whole - * lot of memcpy'ing done in rg_analysis_analyze and why the context - * holds so many buffers. - */ - -#include <math.h> -#include <string.h> -#include <glib.h> - -#include "rganalysis.h" - -#define YULE_ORDER 10 -#define BUTTER_ORDER 2 -/* Percentile which is louder than the proposed level: */ -#define RMS_PERCENTILE 95 -/* Duration of RMS window in milliseconds: */ -#define RMS_WINDOW_MSECS 50 -/* Histogram array elements per dB: */ -#define STEPS_PER_DB 100 -/* Histogram upper bound in dB (normal max. values in the wild are - * assumed to be around 70, 80 dB): */ -#define MAX_DB 120 -/* Calibration value: */ -#define PINK_REF 64.82 /* 298640883795 */ - -#define MAX_ORDER MAX (BUTTER_ORDER, YULE_ORDER) -#define MAX_SAMPLE_RATE 48000 -/* The + 999 has the effect of ceil()ing: */ -#define MAX_SAMPLE_WINDOW (guint) \ - ((MAX_SAMPLE_RATE * RMS_WINDOW_MSECS + 999) / 1000) - -/* Analysis result accumulator. */ - -struct _RgAnalysisAcc -{ - guint32 histogram[STEPS_PER_DB * MAX_DB]; - gdouble peak; -}; - -typedef struct _RgAnalysisAcc RgAnalysisAcc; - -/* Analysis context. */ - -struct _RgAnalysisCtx -{ - /* Filter buffers for left channel. */ - gfloat inprebuf_l[MAX_ORDER * 2]; - gfloat *inpre_l; - gfloat stepbuf_l[MAX_SAMPLE_WINDOW + MAX_ORDER]; - gfloat *step_l; - gfloat outbuf_l[MAX_SAMPLE_WINDOW + MAX_ORDER]; - gfloat *out_l; - /* Filter buffers for right channel. */ - gfloat inprebuf_r[MAX_ORDER * 2]; - gfloat *inpre_r; - gfloat stepbuf_r[MAX_SAMPLE_WINDOW + MAX_ORDER]; - gfloat *step_r; - gfloat outbuf_r[MAX_SAMPLE_WINDOW + MAX_ORDER]; - gfloat *out_r; - - /* Number of samples to reach duration of the RMS window: */ - guint window_n_samples; - /* Progress of the running window: */ - guint window_n_samples_done; - gdouble window_square_sum; - - gint sample_rate; - gint sample_rate_index; - - RgAnalysisAcc track; - RgAnalysisAcc album; -}; - -/* Filter coefficients for the IIR filters that form the equal - * loudness filter. XFilter[ctx->sample_rate_index] gives the array - * of the X coefficients (A or B) for the configured sample rate. */ - -#ifdef _MSC_VER -/* Disable double-to-float warning: */ -/* A better solution would be to append 'f' to each constant, but that - * makes the code ugly. */ -#pragma warning ( disable : 4305 ) -#endif - -static const gfloat AYule[9][11] = { - {1., -3.84664617118067, 7.81501653005538, -11.34170355132042, - 13.05504219327545, -12.28759895145294, 9.48293806319790, - -5.87257861775999, 2.75465861874613, -0.86984376593551, - 0.13919314567432}, - {1., -3.47845948550071, 6.36317777566148, -8.54751527471874, 9.47693607801280, - -8.81498681370155, 6.85401540936998, -4.39470996079559, - 2.19611684890774, -0.75104302451432, 0.13149317958808}, - {1., -2.37898834973084, 2.84868151156327, -2.64577170229825, 2.23697657451713, - -1.67148153367602, 1.00595954808547, -0.45953458054983, - 0.16378164858596, -0.05032077717131, 0.02347897407020}, - {1., -1.61273165137247, 1.07977492259970, -0.25656257754070, - -0.16276719120440, -0.22638893773906, 0.39120800788284, - -0.22138138954925, 0.04500235387352, 0.02005851806501, - 0.00302439095741}, - {1., -1.49858979367799, 0.87350271418188, 0.12205022308084, -0.80774944671438, - 0.47854794562326, -0.12453458140019, -0.04067510197014, - 0.08333755284107, -0.04237348025746, 0.02977207319925}, - {1., -0.62820619233671, 0.29661783706366, -0.37256372942400, 0.00213767857124, - -0.42029820170918, 0.22199650564824, 0.00613424350682, 0.06747620744683, - 0.05784820375801, 0.03222754072173}, - {1., -1.04800335126349, 0.29156311971249, -0.26806001042947, 0.00819999645858, - 0.45054734505008, -0.33032403314006, 0.06739368333110, - -0.04784254229033, 0.01639907836189, 0.01807364323573}, - {1., -0.51035327095184, -0.31863563325245, -0.20256413484477, - 0.14728154134330, 0.38952639978999, -0.23313271880868, - -0.05246019024463, -0.02505961724053, 0.02442357316099, - 0.01818801111503}, - {1., -0.25049871956020, -0.43193942311114, -0.03424681017675, - -0.04678328784242, 0.26408300200955, 0.15113130533216, - -0.17556493366449, -0.18823009262115, 0.05477720428674, - 0.04704409688120} -}; - -static const gfloat BYule[9][11] = { - {0.03857599435200, -0.02160367184185, -0.00123395316851, -0.00009291677959, - -0.01655260341619, 0.02161526843274, -0.02074045215285, - 0.00594298065125, 0.00306428023191, 0.00012025322027, 0.00288463683916}, - {0.05418656406430, -0.02911007808948, -0.00848709379851, -0.00851165645469, - -0.00834990904936, 0.02245293253339, -0.02596338512915, - 0.01624864962975, -0.00240879051584, 0.00674613682247, - -0.00187763777362}, - {0.15457299681924, -0.09331049056315, -0.06247880153653, 0.02163541888798, - -0.05588393329856, 0.04781476674921, 0.00222312597743, 0.03174092540049, - -0.01390589421898, 0.00651420667831, -0.00881362733839}, - {0.30296907319327, -0.22613988682123, -0.08587323730772, 0.03282930172664, - -0.00915702933434, -0.02364141202522, -0.00584456039913, - 0.06276101321749, -0.00000828086748, 0.00205861885564, - -0.02950134983287}, - {0.33642304856132, -0.25572241425570, -0.11828570177555, 0.11921148675203, - -0.07834489609479, -0.00469977914380, -0.00589500224440, - 0.05724228140351, 0.00832043980773, -0.01635381384540, - -0.01760176568150}, - {0.44915256608450, -0.14351757464547, -0.22784394429749, -0.01419140100551, - 0.04078262797139, -0.12398163381748, 0.04097565135648, 0.10478503600251, - -0.01863887810927, -0.03193428438915, 0.00541907748707}, - {0.56619470757641, -0.75464456939302, 0.16242137742230, 0.16744243493672, - -0.18901604199609, 0.30931782841830, -0.27562961986224, - 0.00647310677246, 0.08647503780351, -0.03788984554840, - -0.00588215443421}, - {0.58100494960553, -0.53174909058578, -0.14289799034253, 0.17520704835522, - 0.02377945217615, 0.15558449135573, -0.25344790059353, 0.01628462406333, - 0.06920467763959, -0.03721611395801, -0.00749618797172}, - {0.53648789255105, -0.42163034350696, -0.00275953611929, 0.04267842219415, - -0.10214864179676, 0.14590772289388, -0.02459864859345, - -0.11202315195388, -0.04060034127000, 0.04788665548180, - -0.02217936801134} -}; - -static const gfloat AButter[9][3] = { - {1., -1.97223372919527, 0.97261396931306}, - {1., -1.96977855582618, 0.97022847566350}, - {1., -1.95835380975398, 0.95920349965459}, - {1., -1.95002759149878, 0.95124613669835}, - {1., -1.94561023566527, 0.94705070426118}, - {1., -1.92783286977036, 0.93034775234268}, - {1., -1.91858953033784, 0.92177618768381}, - {1., -1.91542108074780, 0.91885558323625}, - {1., -1.88903307939452, 0.89487434461664} -}; - -static const gfloat BButter[9][3] = { - {0.98621192462708, -1.97242384925416, 0.98621192462708}, - {0.98500175787242, -1.97000351574484, 0.98500175787242}, - {0.97938932735214, -1.95877865470428, 0.97938932735214}, - {0.97531843204928, -1.95063686409857, 0.97531843204928}, - {0.97316523498161, -1.94633046996323, 0.97316523498161}, - {0.96454515552826, -1.92909031105652, 0.96454515552826}, - {0.96009142950541, -1.92018285901082, 0.96009142950541}, - {0.95856916599601, -1.91713833199203, 0.95856916599601}, - {0.94597685600279, -1.89195371200558, 0.94597685600279} -}; - -#ifdef _MSC_VER -#pragma warning ( default : 4305 ) -#endif - -/* Filter functions. These access elements with negative indices of - * the input and output arrays (up to the filter's order). */ - -/* For much better performance, the function below has been - * implemented by unrolling the inner loop for our two use cases. */ - -/* - * static inline void - * apply_filter (const gfloat * input, gfloat * output, guint n_samples, - * const gfloat * a, const gfloat * b, guint order) - * { - * gfloat y; - * gint i, k; - * - * for (i = 0; i < n_samples; i++) { - * y = input[i] * b[0]; - * for (k = 1; k <= order; k++) - * y += input[i - k] * b[k] - output[i - k] * a[k]; - * output[i] = y; - * } - * } - */ - -static inline void -yule_filter (const gfloat * input, gfloat * output, - const gfloat * a, const gfloat * b) -{ - /* 1e-10 is added below to avoid running into denormals when operating on - * near silence. */ - - output[0] = 1e-10 + input[0] * b[0] - + input[-1] * b[1] - output[-1] * a[1] - + input[-2] * b[2] - output[-2] * a[2] - + input[-3] * b[3] - output[-3] * a[3] - + input[-4] * b[4] - output[-4] * a[4] - + input[-5] * b[5] - output[-5] * a[5] - + input[-6] * b[6] - output[-6] * a[6] - + input[-7] * b[7] - output[-7] * a[7] - + input[-8] * b[8] - output[-8] * a[8] - + input[-9] * b[9] - output[-9] * a[9] - + input[-10] * b[10] - output[-10] * a[10]; -} - -static inline void -butter_filter (const gfloat * input, gfloat * output, - const gfloat * a, const gfloat * b) -{ - output[0] = input[0] * b[0] - + input[-1] * b[1] - output[-1] * a[1] - + input[-2] * b[2] - output[-2] * a[2]; -} - -/* Because butter_filter and yule_filter are inlined, this function is - * a bit blown-up (code-size wise), but not inlining gives a ca. 40% - * performance penalty. */ - -static inline void -apply_filters (const RgAnalysisCtx * ctx, const gfloat * input_l, - const gfloat * input_r, guint n_samples) -{ - const gfloat *ayule = AYule[ctx->sample_rate_index]; - const gfloat *byule = BYule[ctx->sample_rate_index]; - const gfloat *abutter = AButter[ctx->sample_rate_index]; - const gfloat *bbutter = BButter[ctx->sample_rate_index]; - gint pos = ctx->window_n_samples_done; - gint i; - - for (i = 0; i < n_samples; i++, pos++) { - yule_filter (input_l + i, ctx->step_l + pos, ayule, byule); - butter_filter (ctx->step_l + pos, ctx->out_l + pos, abutter, bbutter); - - yule_filter (input_r + i, ctx->step_r + pos, ayule, byule); - butter_filter (ctx->step_r + pos, ctx->out_r + pos, abutter, bbutter); - } -} - -/* Clear filter buffer state and current RMS window. */ - -static void -reset_filters (RgAnalysisCtx * ctx) -{ - gint i; - - for (i = 0; i < MAX_ORDER; i++) { - - ctx->inprebuf_l[i] = 0.; - ctx->stepbuf_l[i] = 0.; - ctx->outbuf_l[i] = 0.; - - ctx->inprebuf_r[i] = 0.; - ctx->stepbuf_r[i] = 0.; - ctx->outbuf_r[i] = 0.; - } - - ctx->window_square_sum = 0.; - ctx->window_n_samples_done = 0; -} - -/* Accumulator functions. */ - -/* Add two accumulators in-place. The sum is defined as the result of - * the vector sum of the histogram array and the maximum value of the - * peak field. Thus "adding" the accumulators for all tracks yields - * the correct result for obtaining the album gain and peak. */ - -static void -accumulator_add (RgAnalysisAcc * acc, const RgAnalysisAcc * acc_other) -{ - gint i; - - for (i = 0; i < G_N_ELEMENTS (acc->histogram); i++) - acc->histogram[i] += acc_other->histogram[i]; - - acc->peak = MAX (acc->peak, acc_other->peak); -} - -/* Reset an accumulator to zero. */ - -static void -accumulator_clear (RgAnalysisAcc * acc) -{ - memset (acc->histogram, 0, sizeof (acc->histogram)); - acc->peak = 0.; -} - -/* Obtain final analysis result from an accumulator. Returns TRUE on - * success, FALSE on error (if accumulator is still zero). */ - -static gboolean -accumulator_result (const RgAnalysisAcc * acc, gdouble * result_gain, - gdouble * result_peak) -{ - guint32 sum = 0; - guint32 upper; - guint i; - - for (i = 0; i < G_N_ELEMENTS (acc->histogram); i++) - sum += acc->histogram[i]; - - if (sum == 0) - /* All entries are 0: We got less than 50ms of data. */ - return FALSE; - - upper = (guint32) ceil (sum * (1. - (gdouble) (RMS_PERCENTILE / 100.))); - - for (i = G_N_ELEMENTS (acc->histogram); i--;) { - if (upper <= acc->histogram[i]) - break; - upper -= acc->histogram[i]; - } - - if (result_peak != NULL) - *result_peak = acc->peak; - if (result_gain != NULL) - *result_gain = PINK_REF - (gdouble) i / STEPS_PER_DB; - - return TRUE; -} - -/* Functions that operate on contexts, for external usage. */ - -/* Create a new context. Before it can be used, a sample rate must be - * configured using rg_analysis_set_sample_rate. */ - -RgAnalysisCtx * -rg_analysis_new (void) -{ - RgAnalysisCtx *ctx; - - ctx = g_new (RgAnalysisCtx, 1); - - ctx->inpre_l = ctx->inprebuf_l + MAX_ORDER; - ctx->step_l = ctx->stepbuf_l + MAX_ORDER; - ctx->out_l = ctx->outbuf_l + MAX_ORDER; - - ctx->inpre_r = ctx->inprebuf_r + MAX_ORDER; - ctx->step_r = ctx->stepbuf_r + MAX_ORDER; - ctx->out_r = ctx->outbuf_r + MAX_ORDER; - - ctx->sample_rate = 0; - - accumulator_clear (&ctx->track); - accumulator_clear (&ctx->album); - - return ctx; -} - -/* Adapt to given sample rate. Does nothing if already the current - * rate (returns TRUE then). Returns FALSE only if given sample rate - * is not supported. If the configured rate changes, the last - * unprocessed incomplete 50ms chunk of data is dropped because the - * filters are reset. */ - -gboolean -rg_analysis_set_sample_rate (RgAnalysisCtx * ctx, gint sample_rate) -{ - g_return_val_if_fail (ctx != NULL, FALSE); - - if (ctx->sample_rate == sample_rate) - return TRUE; - - switch (sample_rate) { - case 48000: - ctx->sample_rate_index = 0; - break; - case 44100: - ctx->sample_rate_index = 1; - break; - case 32000: - ctx->sample_rate_index = 2; - break; - case 24000: - ctx->sample_rate_index = 3; - break; - case 22050: - ctx->sample_rate_index = 4; - break; - case 16000: - ctx->sample_rate_index = 5; - break; - case 12000: - ctx->sample_rate_index = 6; - break; - case 11025: - ctx->sample_rate_index = 7; - break; - case 8000: - ctx->sample_rate_index = 8; - break; - default: - return FALSE; - } - - ctx->sample_rate = sample_rate; - /* The + 999 has the effect of ceil()ing: */ - ctx->window_n_samples = (guint) ((sample_rate * RMS_WINDOW_MSECS + 999) - / 1000); - - reset_filters (ctx); - - return TRUE; -} - -void -rg_analysis_destroy (RgAnalysisCtx * ctx) -{ - g_free (ctx); -} - -/* Entry points for analyzing sample data in common raw data formats. - * The stereo format functions expect interleaved frames. It is - * possible to pass data in different formats for the same context, - * there are no restrictions. All functions have the same signature; - * the depth argument for the float functions is not variable and must - * be given the value 32. */ - -void -rg_analysis_analyze_mono_float (RgAnalysisCtx * ctx, gconstpointer data, - gsize size, guint depth) -{ - gfloat conv_samples[512]; - const gfloat *samples = (gfloat *) data; - guint n_samples = size / sizeof (gfloat); - gint i; - - g_return_if_fail (depth == 32); - g_return_if_fail (size % sizeof (gfloat) == 0); - - while (n_samples) { - gint n = MIN (n_samples, G_N_ELEMENTS (conv_samples)); - - n_samples -= n; - memcpy (conv_samples, samples, n * sizeof (gfloat)); - for (i = 0; i < n; i++) { - ctx->track.peak = MAX (ctx->track.peak, fabs (conv_samples[i])); - conv_samples[i] *= 32768.; - } - samples += n; - rg_analysis_analyze (ctx, conv_samples, NULL, n); - } -} - -void -rg_analysis_analyze_stereo_float (RgAnalysisCtx * ctx, gconstpointer data, - gsize size, guint depth) -{ - gfloat conv_samples_l[256]; - gfloat conv_samples_r[256]; - const gfloat *samples = (gfloat *) data; - guint n_frames = size / (sizeof (gfloat) * 2); - gint i; - - g_return_if_fail (depth == 32); - g_return_if_fail (size % (sizeof (gfloat) * 2) == 0); - - while (n_frames) { - gint n = MIN (n_frames, G_N_ELEMENTS (conv_samples_l)); - - n_frames -= n; - for (i = 0; i < n; i++) { - gfloat old_sample; - - old_sample = samples[2 * i]; - ctx->track.peak = MAX (ctx->track.peak, fabs (old_sample)); - conv_samples_l[i] = old_sample * 32768.; - - old_sample = samples[2 * i + 1]; - ctx->track.peak = MAX (ctx->track.peak, fabs (old_sample)); - conv_samples_r[i] = old_sample * 32768.; - } - samples += 2 * n; - rg_analysis_analyze (ctx, conv_samples_l, conv_samples_r, n); - } -} - -void -rg_analysis_analyze_mono_int16 (RgAnalysisCtx * ctx, gconstpointer data, - gsize size, guint depth) -{ - gfloat conv_samples[512]; - gint32 peak_sample = 0; - const gint16 *samples = (gint16 *) data; - guint n_samples = size / sizeof (gint16); - gint shift = sizeof (gint16) * 8 - depth; - gint i; - - g_return_if_fail (depth <= (sizeof (gint16) * 8)); - g_return_if_fail (size % sizeof (gint16) == 0); - - while (n_samples) { - gint n = MIN (n_samples, G_N_ELEMENTS (conv_samples)); - - n_samples -= n; - for (i = 0; i < n; i++) { - gint16 old_sample = samples[i] << shift; - - peak_sample = MAX (peak_sample, ABS ((gint32) old_sample)); - conv_samples[i] = (gfloat) old_sample; - } - samples += n; - rg_analysis_analyze (ctx, conv_samples, NULL, n); - } - ctx->track.peak = MAX (ctx->track.peak, - (gdouble) peak_sample / ((gdouble) (1u << 15))); -} - -void -rg_analysis_analyze_stereo_int16 (RgAnalysisCtx * ctx, gconstpointer data, - gsize size, guint depth) -{ - gfloat conv_samples_l[256]; - gfloat conv_samples_r[256]; - gint32 peak_sample = 0; - const gint16 *samples = (gint16 *) data; - guint n_frames = size / (sizeof (gint16) * 2); - gint shift = sizeof (gint16) * 8 - depth; - gint i; - - g_return_if_fail (depth <= (sizeof (gint16) * 8)); - g_return_if_fail (size % (sizeof (gint16) * 2) == 0); - - while (n_frames) { - gint n = MIN (n_frames, G_N_ELEMENTS (conv_samples_l)); - - n_frames -= n; - for (i = 0; i < n; i++) { - gint16 old_sample; - - old_sample = samples[2 * i] << shift; - peak_sample = MAX (peak_sample, ABS ((gint32) old_sample)); - conv_samples_l[i] = (gfloat) old_sample; - - old_sample = samples[2 * i + 1] << shift; - peak_sample = MAX (peak_sample, ABS ((gint32) old_sample)); - conv_samples_r[i] = (gfloat) old_sample; - } - samples += 2 * n; - rg_analysis_analyze (ctx, conv_samples_l, conv_samples_r, n); - } - ctx->track.peak = MAX (ctx->track.peak, - (gdouble) peak_sample / ((gdouble) (1u << 15))); -} - -/* Analyze the given chunk of samples. The sample data is given in - * floating point format but should be scaled such that the values - * +/-32768.0 correspond to the -0dBFS reference amplitude. - * - * samples_l: Buffer with sample data for the left channel or of the - * mono channel. - * - * samples_r: Buffer with sample data for the right channel or NULL - * for mono. - * - * n_samples: Number of samples passed in each buffer. - */ - -void -rg_analysis_analyze (RgAnalysisCtx * ctx, const gfloat * samples_l, - const gfloat * samples_r, guint n_samples) -{ - const gfloat *input_l, *input_r; - guint n_samples_done; - gint i; - - g_return_if_fail (ctx != NULL); - g_return_if_fail (samples_l != NULL); - g_return_if_fail (ctx->sample_rate != 0); - - if (n_samples == 0) - return; - - if (samples_r == NULL) - /* Mono. */ - samples_r = samples_l; - - memcpy (ctx->inpre_l, samples_l, - MIN (n_samples, MAX_ORDER) * sizeof (gfloat)); - memcpy (ctx->inpre_r, samples_r, - MIN (n_samples, MAX_ORDER) * sizeof (gfloat)); - - n_samples_done = 0; - while (n_samples_done < n_samples) { - /* Limit number of samples to be processed in this iteration to - * the number needed to complete the next window: */ - guint n_samples_current = MIN (n_samples - n_samples_done, - ctx->window_n_samples - ctx->window_n_samples_done); - - if (n_samples_done < MAX_ORDER) { - input_l = ctx->inpre_l + n_samples_done; - input_r = ctx->inpre_r + n_samples_done; - n_samples_current = MIN (n_samples_current, MAX_ORDER - n_samples_done); - } else { - input_l = samples_l + n_samples_done; - input_r = samples_r + n_samples_done; - } - - apply_filters (ctx, input_l, input_r, n_samples_current); - - /* Update the square sum. */ - for (i = 0; i < n_samples_current; i++) - ctx->window_square_sum += ctx->out_l[ctx->window_n_samples_done + i] - * ctx->out_l[ctx->window_n_samples_done + i] - + ctx->out_r[ctx->window_n_samples_done + i] - * ctx->out_r[ctx->window_n_samples_done + i]; - - ctx->window_n_samples_done += n_samples_current; - - g_return_if_fail (ctx->window_n_samples_done <= ctx->window_n_samples); - - if (ctx->window_n_samples_done == ctx->window_n_samples) { - /* Get the Root Mean Square (RMS) for this set of samples. */ - gdouble val = STEPS_PER_DB * 10. * log10 (ctx->window_square_sum / - ctx->window_n_samples * 0.5 + 1.e-37); - gint ival = CLAMP ((gint) val, 0, - (gint) G_N_ELEMENTS (ctx->track.histogram) - 1); - - ctx->track.histogram[ival]++; - ctx->window_square_sum = 0.; - ctx->window_n_samples_done = 0; - - /* No need for memmove here, the areas never overlap: Even for - * the smallest sample rate, the number of samples needed for - * the window is greater than MAX_ORDER. */ - - memcpy (ctx->stepbuf_l, ctx->stepbuf_l + ctx->window_n_samples, - MAX_ORDER * sizeof (gfloat)); - memcpy (ctx->outbuf_l, ctx->outbuf_l + ctx->window_n_samples, - MAX_ORDER * sizeof (gfloat)); - - memcpy (ctx->stepbuf_r, ctx->stepbuf_r + ctx->window_n_samples, - MAX_ORDER * sizeof (gfloat)); - memcpy (ctx->outbuf_r, ctx->outbuf_r + ctx->window_n_samples, - MAX_ORDER * sizeof (gfloat)); - } - - n_samples_done += n_samples_current; - } - - if (n_samples >= MAX_ORDER) { - - memcpy (ctx->inprebuf_l, samples_l + n_samples - MAX_ORDER, - MAX_ORDER * sizeof (gfloat)); - - memcpy (ctx->inprebuf_r, samples_r + n_samples - MAX_ORDER, - MAX_ORDER * sizeof (gfloat)); - - } else { - - memmove (ctx->inprebuf_l, ctx->inprebuf_l + n_samples, - (MAX_ORDER - n_samples) * sizeof (gfloat)); - memcpy (ctx->inprebuf_l + MAX_ORDER - n_samples, samples_l, - n_samples * sizeof (gfloat)); - - memmove (ctx->inprebuf_r, ctx->inprebuf_r + n_samples, - (MAX_ORDER - n_samples) * sizeof (gfloat)); - memcpy (ctx->inprebuf_r + MAX_ORDER - n_samples, samples_r, - n_samples * sizeof (gfloat)); - - } -} - -/* Obtain track gain and peak. Returns TRUE on success. Can fail if - * not enough samples have been processed. Updates album accumulator. - * Resets track accumulator. */ - -gboolean -rg_analysis_track_result (RgAnalysisCtx * ctx, gdouble * gain, gdouble * peak) -{ - gboolean result; - - g_return_val_if_fail (ctx != NULL, FALSE); - - accumulator_add (&ctx->album, &ctx->track); - result = accumulator_result (&ctx->track, gain, peak); - accumulator_clear (&ctx->track); - - reset_filters (ctx); - - return result; -} - -/* Obtain album gain and peak. Returns TRUE on success. Can fail if - * not enough samples have been processed. Resets album - * accumulator. */ - -gboolean -rg_analysis_album_result (RgAnalysisCtx * ctx, gdouble * gain, gdouble * peak) -{ - gboolean result; - - g_return_val_if_fail (ctx != NULL, FALSE); - - result = accumulator_result (&ctx->album, gain, peak); - accumulator_clear (&ctx->album); - - return result; -} - -void -rg_analysis_reset_album (RgAnalysisCtx * ctx) -{ - accumulator_clear (&ctx->album); -} - -/* Reset internal buffers as well as track and album accumulators. - * Configured sample rate is kept intact. */ - -void -rg_analysis_reset (RgAnalysisCtx * ctx) -{ - g_return_if_fail (ctx != NULL); - - reset_filters (ctx); - accumulator_clear (&ctx->track); - accumulator_clear (&ctx->album); -} diff --git a/gst/replaygain/rganalysis.h b/gst/replaygain/rganalysis.h deleted file mode 100644 index 16247361..00000000 --- a/gst/replaygain/rganalysis.h +++ /dev/null @@ -1,56 +0,0 @@ -/* GStreamer ReplayGain analysis - * - * Copyright (C) 2006 Rene Stadler <mail@renestadler.de> - * Copyright (C) 2001 David Robinson <David@Robinson.org> - * Glen Sawyer <glensawyer@hotmail.com> - * - * rganalysis.h: Analyze raw audio data in accordance with ReplayGain - * - * This library is free software; you can redistribute it and/or - * modify it under the terms of the GNU Lesser General Public License - * as published by the Free Software Foundation; either version 2.1 of - * the License, or (at your option) any later version. - * - * This library is distributed in the hope that it will be useful, but - * WITHOUT ANY WARRANTY; without even the implied warranty of - * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU - * Lesser General Public License for more details. - * - * You should have received a copy of the GNU Lesser General Public - * License along with this library; if not, write to the Free Software - * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA - * 02110-1301 USA - */ - -#ifndef __RG_ANALYSIS_H__ -#define __RG_ANALYSIS_H__ - -#include <glib.h> - -G_BEGIN_DECLS - -typedef struct _RgAnalysisCtx RgAnalysisCtx; - -RgAnalysisCtx *rg_analysis_new (void); -gboolean rg_analysis_set_sample_rate (RgAnalysisCtx * ctx, gint sample_rate); -void rg_analysis_analyze_mono_float (RgAnalysisCtx * ctx, gconstpointer data, - gsize size, guint depth); -void rg_analysis_analyze_stereo_float (RgAnalysisCtx * ctx, gconstpointer data, - gsize size, guint depth); -void rg_analysis_analyze_mono_int16 (RgAnalysisCtx * ctx, gconstpointer data, - gsize size, guint depth); -void rg_analysis_analyze_stereo_int16 (RgAnalysisCtx * ctx, gconstpointer data, - gsize size, guint depth); -void rg_analysis_analyze (RgAnalysisCtx * ctx, const gfloat * samples_l, - const gfloat * samples_r, guint n_samples); -gboolean rg_analysis_track_result (RgAnalysisCtx * ctx, gdouble * gain, - gdouble * peak); -gboolean rg_analysis_album_result (RgAnalysisCtx * ctx, gdouble * gain, - gdouble * peak); -void rg_analysis_reset_album (RgAnalysisCtx * ctx); -void rg_analysis_reset (RgAnalysisCtx * ctx); -void rg_analysis_destroy (RgAnalysisCtx * ctx); - -G_END_DECLS - -#endif /* __RG_ANALYSIS_H__ */ diff --git a/tests/check/Makefile.am b/tests/check/Makefile.am index 6cc8163e..24070fd8 100644 --- a/tests/check/Makefile.am +++ b/tests/check/Makefile.am @@ -73,11 +73,6 @@ check_PROGRAMS = \ $(check_neon) \ $(check_ofa) \ $(check_timidity) \ - elements/deinterleave \ - elements/interleave \ - elements/rganalysis \ - elements/rglimiter \ - elements/rgvolume \ elements/selector \ elements/y4menc @@ -88,11 +83,6 @@ TESTS = $(check_PROGRAMS) 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) diff --git a/tests/check/elements/deinterleave.c b/tests/check/elements/deinterleave.c deleted file mode 100644 index 04ac41b3..00000000 --- a/tests/check/elements/deinterleave.c +++ /dev/null @@ -1,558 +0,0 @@ -/* 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); diff --git a/tests/check/elements/interleave.c b/tests/check/elements/interleave.c deleted file mode 100644 index 6b476046..00000000 --- a/tests/check/elements/interleave.c +++ /dev/null @@ -1,761 +0,0 @@ -/* 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); diff --git a/tests/check/elements/rganalysis.c b/tests/check/elements/rganalysis.c deleted file mode 100644 index 0045cb94..00000000 --- a/tests/check/elements/rganalysis.c +++ /dev/null @@ -1,1925 +0,0 @@ -/* 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; -} diff --git a/tests/check/elements/rglimiter.c b/tests/check/elements/rglimiter.c deleted file mode 100644 index 9d838785..00000000 --- a/tests/check/elements/rglimiter.c +++ /dev/null @@ -1,268 +0,0 @@ -/* 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; -} diff --git a/tests/check/elements/rgvolume.c b/tests/check/elements/rgvolume.c deleted file mode 100644 index 7159bb76..00000000 --- a/tests/check/elements/rgvolume.c +++ /dev/null @@ -1,573 +0,0 @@ -/* 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; -} |