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Remove interleave and replaygain plugins that have moved to -good

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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
This commit is contained in:
Jan Schmidt 2008-07-19 00:58:49 +00:00
parent 26cb95316c
commit e985585a4e
31 changed files with 40 additions and 9573 deletions
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37
ChangeLog
View file

@ -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:

5
docs/plugins/Makefile.am
View file

@ -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 \

7
docs/plugins/gst-plugins-bad-plugins-docs.sgml
View file

@ -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" />

75
docs/plugins/gst-plugins-bad-plugins-sections.txt
View file

@ -168,6 +168,7 @@ FESTIVAL_DEFAULT_SERVER_PORT
FESTIVAL_DEFAULT_TEXT_MODE
</SECTION>
<SECTION>
<FILE>element-input-selector</FILE>
<TITLE>input-selector</TITLE>
GstInputSelector
@ -201,38 +202,6 @@ GST_TYPE_IVORBIS_DEC
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
@ -380,48 +349,6 @@ GST_TYPE_OUTPUT_SELECTOR
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>

2
docs/plugins/gst-plugins-bad-plugins.prerequisites
View file

@ -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

55
docs/plugins/inspect/plugin-interleave.xml
View file

@ -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 &lt;wingo at pobox.com&gt;, Iain &lt;iain@prettypeople.org&gt;, Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</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 &lt;wingo at pobox.com&gt;, Sebastian Dröge &lt;slomo@circular-chaos.org&gt;</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>

76
docs/plugins/inspect/plugin-replaygain.xml
View file

@ -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 &lt;mail@renestadler.de&gt;</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 &lt;mail@renestadler.de&gt;</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 &lt;mail@renestadler.de&gt;</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>

9
gst/interleave/Makefile.am
View file

@ -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

889
gst/interleave/deinterleave.c
View file

@ -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;
}

75
gst/interleave/deinterleave.h
View file

@ -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__ */

1352
gst/interleave/interleave.c
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89
gst/interleave/interleave.h
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/* 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__ */

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gst/interleave/plugin.c
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/* 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);

31
gst/interleave/plugin.h
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/* 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__ */

21
gst/replaygain/Makefile.am
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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

692
gst/replaygain/gstrganalysis.c
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/* 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;
}

85
gst/replaygain/gstrganalysis.h
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/* 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__ */

202
gst/replaygain/gstrglimiter.c
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@ -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;
}

64
gst/replaygain/gstrglimiter.h
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@ -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__ */

698
gst/replaygain/gstrgvolume.c
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@ -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;
}
}

88
gst/replaygain/gstrgvolume.h
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@ -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__ */

53
gst/replaygain/replaygain.c
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@ -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);

36
gst/replaygain/replaygain.h
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@ -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__ */

777
gst/replaygain/rganalysis.c
View file

@ -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);
}

56
gst/replaygain/rganalysis.h
View file

@ -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__ */

10
tests/check/Makefile.am
View file

@ -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)

558
tests/check/elements/deinterleave.c
View file

@ -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);

761
tests/check/elements/interleave.c
View file

@ -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);

1925
tests/check/elements/rganalysis.c
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File diff suppressed because it is too large Load diff

268
tests/check/elements/rglimiter.c
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@ -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;
}

573
tests/check/elements/rgvolume.c
View file

@ -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;
}