gstreamer/gst/adder/gstadder.c

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/* GStreamer
* Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu>
* 2001 Thomas <thomas@apestaart.org>
* 2005,2006 Wim Taymans <wim@fluendo.com>
*
* adder.c: Adder element, N in, one out, samples are added
*
* 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.
*/
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
/**
* SECTION:element-adder
*
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
2008-07-10 21:06:06 +00:00
* The adder allows to mix several streams into one by adding the data.
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
* Mixed data is clamped to the min/max values of the data format.
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
2008-07-10 21:06:06 +00:00
*
* The adder currently mixes all data received on the sinkpads as soon as
* possible without trying to synchronize the streams.
*
* <refsect2>
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
* <title>Example launch line</title>
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
2008-07-10 21:06:06 +00:00
* |[
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
* gst-launch audiotestsrc freq=100 ! adder name=mix ! audioconvert ! alsasink audiotestsrc freq=500 ! mix.
Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipe... Original commit message from CVS: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: * docs/plugins/gst-plugins-base-plugins.args: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: * docs/plugins/gst-plugins-base-plugins.signals: * docs/plugins/inspect/plugin-adder.xml: * docs/plugins/inspect/plugin-alsa.xml: * docs/plugins/inspect/plugin-audioconvert.xml: * docs/plugins/inspect/plugin-audiorate.xml: * docs/plugins/inspect/plugin-audioresample.xml: * docs/plugins/inspect/plugin-audiotestsrc.xml: * docs/plugins/inspect/plugin-cdparanoia.xml: * docs/plugins/inspect/plugin-decodebin.xml: * docs/plugins/inspect/plugin-ffmpegcolorspace.xml: * docs/plugins/inspect/plugin-gdp.xml: * docs/plugins/inspect/plugin-gnomevfs.xml: * docs/plugins/inspect/plugin-libvisual.xml: * docs/plugins/inspect/plugin-ogg.xml: * docs/plugins/inspect/plugin-pango.xml: * docs/plugins/inspect/plugin-playback.xml: * docs/plugins/inspect/plugin-queue2.xml: * docs/plugins/inspect/plugin-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.xml: * docs/plugins/inspect/plugin-uridecodebin.xml: * docs/plugins/inspect/plugin-video4linux.xml: * docs/plugins/inspect/plugin-videorate.xml: * docs/plugins/inspect/plugin-videoscale.xml: * docs/plugins/inspect/plugin-videotestsrc.xml: * docs/plugins/inspect/plugin-volume.xml: * docs/plugins/inspect/plugin-vorbis.xml: * docs/plugins/inspect/plugin-ximagesink.xml: * docs/plugins/inspect/plugin-xvimagesink.xml: * ext/alsa/gstalsamixer.c: * ext/alsa/gstalsasink.c: * ext/alsa/gstalsasrc.c: * ext/gio/gstgiosink.c: * ext/gio/gstgiosrc.c: * ext/gio/gstgiostreamsink.c: * ext/gio/gstgiostreamsrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/theora/theoraparse.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * ext/vorbis/vorbisparse.c: * ext/vorbis/vorbistag.c: * gst/adder/gstadder.c: * gst/audioconvert/gstaudioconvert.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/gdp/gstgdpdepay.c: * gst/gdp/gstgdppay.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gstqueue2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpserversink.c: * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: Cleanup Plugin docs. Link to signals and properties. Fix sub-section titles. Drop mentining that all our example pipelines are "simple" pipelines.
2008-07-10 21:06:06 +00:00
* ]| This pipeline produces two sine waves mixed together.
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
* </refsect2>
*
* Last reviewed on 2006-05-09 (0.10.7)
Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_base_init), (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_base_init): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_base_init): * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: (gst_ogg_parse_base_init): * ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_base_init), (gst_ogm_video_parse_base_init), (gst_ogm_text_parse_base_init): * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_base_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_base_init): * gst/adder/gstadder.c: (gst_adder_get_type): * gst/adder/gstadder.h: * gst/audioconvert/gstaudioconvert.c: * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audiostestsrc_wave_get_type), (gst_audio_test_src_class_init), (gst_audio_test_src_create): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: (gst_stream_selector_base_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_base_init): * gst/volume/gstvolume.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * tests/check/libs/cddabasesrc.c: * tests/old/examples/gob/gst-identity2.gob: Add docs for adder, use GST_ELEMENT_DETAILS macro, define GstElementDetails at the top
2006-03-24 10:42:11 +00:00
*/
/* Element-Checklist-Version: 5 */
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstadder.h"
#include <gst/audio/audio.h>
#include <string.h> /* strcmp */
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#include "gstadderorc.h"
/* highest positive/lowest negative x-bit value we can use for clamping */
#define MAX_INT_32 ((gint32) (0x7fffffff))
#define MAX_INT_16 ((gint16) (0x7fff))
#define MAX_INT_8 ((gint8) (0x7f))
#define MAX_UINT_32 ((guint32)(0xffffffff))
#define MAX_UINT_16 ((guint16)(0xffff))
#define MAX_UINT_8 ((guint8) (0xff))
#define MIN_INT_32 ((gint32) (0x80000000))
#define MIN_INT_16 ((gint16) (0x8000))
#define MIN_INT_8 ((gint8) (0x80))
#define MIN_UINT_32 ((guint32)(0x00000000))
#define MIN_UINT_16 ((guint16)(0x0000))
#define MIN_UINT_8 ((guint8) (0x00))
enum
{
PROP_0,
PROP_FILTER_CAPS
};
#define GST_CAT_DEFAULT gst_adder_debug
GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT);
/* elementfactory information */
#if G_BYTE_ORDER == G_LITTLE_ENDIAN
#define CAPS \
GST_AUDIO_CAPS_MAKE ("{ S32LE, U32LE, S16LE, U16LE, S8, U8, F32LE, F64LE }") \
", layout = (string) { interleaved, non-interleaved }"
#else
#define CAPS \
GST_AUDIO_CAPS_MAKE ("{ S32BE, U32BE, S16BE, U16BE, S8, U8, F32BE, F64BE }") \
", layout = (string) { interleaved, non-interleaved }"
#endif
static GstStaticPadTemplate gst_adder_src_template =
GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS (CAPS)
);
static GstStaticPadTemplate gst_adder_sink_template =
GST_STATIC_PAD_TEMPLATE ("sink_%u",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS (CAPS)
);
#define gst_adder_parent_class parent_class
G_DEFINE_TYPE (GstAdder, gst_adder, GST_TYPE_ELEMENT);
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static void gst_adder_dispose (GObject * object);
static void gst_adder_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec);
static void gst_adder_get_property (GObject * object, guint prop_id,
GValue * value, GParamSpec * pspec);
static gboolean gst_adder_setcaps (GstAdder * adder, GstPad * pad,
GstCaps * caps);
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static gboolean gst_adder_src_query (GstPad * pad, GstObject * parent,
GstQuery * query);
static gboolean gst_adder_sink_query (GstPad * pad, GstObject * parent,
GstQuery * query);
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static gboolean gst_adder_src_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static gboolean gst_adder_sink_event (GstPad * pad, GstObject * parent,
GstEvent * event);
static GstPad *gst_adder_request_new_pad (GstElement * element,
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GstPadTemplate * temp, const gchar * unused, const GstCaps * caps);
static void gst_adder_release_pad (GstElement * element, GstPad * pad);
static GstStateChangeReturn gst_adder_change_state (GstElement * element,
GstStateChange transition);
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static GstFlowReturn gst_adder_do_clip (GstCollectPads2 * pads,
GstCollectData2 * data, GstBuffer * buffer, GstBuffer ** out,
gpointer user_data);
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static GstFlowReturn gst_adder_collected (GstCollectPads2 * pads,
gpointer user_data);
static gboolean gst_adder_event (GstCollectPads2 * pads, GstCollectData2 * pad,
GstEvent * event, gpointer user_data);
/* non-clipping versions (for float) */
#define MAKE_FUNC_NC(name,type) \
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static void name (type *out, type *in, gint samples) { \
gint i; \
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for (i = 0; i < samples; i++) \
out[i] += in[i]; \
}
/* *INDENT-OFF* */
MAKE_FUNC_NC (add_float64, gdouble)
/* *INDENT-ON* */
/* we can only accept caps that we and downstream can handle.
* if we have filtercaps set, use those to constrain the target caps.
*/
static GstCaps *
gst_adder_sink_getcaps (GstPad * pad, GstCaps * filter)
{
GstAdder *adder;
GstCaps *result, *peercaps, *sinkcaps, *filter_caps;
adder = GST_ADDER (GST_PAD_PARENT (pad));
GST_OBJECT_LOCK (adder);
/* take filter */
if ((filter_caps = adder->filter_caps)) {
if (filter)
filter_caps =
gst_caps_intersect_full (filter, filter_caps,
GST_CAPS_INTERSECT_FIRST);
else
gst_caps_ref (filter_caps);
} else {
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filter_caps = filter ? gst_caps_ref (filter) : NULL;
}
GST_OBJECT_UNLOCK (adder);
if (filter_caps && gst_caps_is_empty (filter_caps)) {
GST_WARNING_OBJECT (pad, "Empty filter caps");
return filter_caps;
}
/* get the downstream possible caps */
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peercaps = gst_pad_peer_query_caps (adder->srcpad, filter_caps);
/* get the allowed caps on this sinkpad */
sinkcaps = gst_pad_get_current_caps (pad);
if (sinkcaps == NULL) {
sinkcaps = gst_pad_get_pad_template_caps (pad);
if (!sinkcaps)
sinkcaps = gst_caps_new_any ();
}
if (peercaps) {
/* if the peer has caps, intersect */
GST_DEBUG_OBJECT (adder, "intersecting peer and template caps");
result =
gst_caps_intersect_full (peercaps, sinkcaps, GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (peercaps);
gst_caps_unref (sinkcaps);
} else {
/* the peer has no caps (or there is no peer), just use the allowed caps
* of this sinkpad. */
/* restrict with filter-caps if any */
if (filter_caps) {
GST_DEBUG_OBJECT (adder, "no peer caps, using filtered sinkcaps");
result =
gst_caps_intersect_full (filter_caps, sinkcaps,
GST_CAPS_INTERSECT_FIRST);
gst_caps_unref (sinkcaps);
} else {
GST_DEBUG_OBJECT (adder, "no peer caps, using sinkcaps");
result = sinkcaps;
}
}
if (filter_caps)
gst_caps_unref (filter_caps);
GST_LOG_OBJECT (adder, "getting caps on pad %p,%s to %" GST_PTR_FORMAT, pad,
GST_PAD_NAME (pad), result);
return result;
}
static gboolean
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gst_adder_sink_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_CAPS:
{
GstCaps *filter, *caps;
gst_query_parse_caps (query, &filter);
caps = gst_adder_sink_getcaps (pad, filter);
gst_query_set_caps_result (query, caps);
gst_caps_unref (caps);
res = TRUE;
break;
}
default:
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res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
typedef struct
{
GstPad *pad;
GstCaps *caps;
} IterData;
static void
setcapsfunc (const GValue * item, IterData * data)
{
GstPad *otherpad = g_value_get_object (item);
if (otherpad != data->pad)
gst_pad_set_caps (data->pad, data->caps);
}
/* the first caps we receive on any of the sinkpads will define the caps for all
* the other sinkpads because we can only mix streams with the same caps.
*/
static gboolean
gst_adder_setcaps (GstAdder * adder, GstPad * pad, GstCaps * caps)
{
GstIterator *it;
GstIteratorResult ires;
IterData idata;
gboolean done;
/* this get called recursively due to gst_iterator_foreach calling
* gst_pad_set_caps() */
if (adder->in_setcaps)
return TRUE;
GST_LOG_OBJECT (adder, "setting caps pad %p,%s to %" GST_PTR_FORMAT, pad,
GST_PAD_NAME (pad), caps);
it = gst_element_iterate_pads (GST_ELEMENT_CAST (adder));
/* FIXME, see if the other pads can accept the format. Also lock the
* format on the other pads to this new format. */
idata.caps = caps;
idata.pad = pad;
adder->in_setcaps = TRUE;
done = FALSE;
while (!done) {
ires = gst_iterator_foreach (it, (GstIteratorForeachFunction) setcapsfunc,
&idata);
switch (ires) {
case GST_ITERATOR_RESYNC:
gst_iterator_resync (it);
break;
default:
done = TRUE;
break;
}
}
adder->in_setcaps = FALSE;
GST_LOG_OBJECT (adder, "handle caps changes on pad %p,%s to %" GST_PTR_FORMAT,
pad, GST_PAD_NAME (pad), caps);
if (!gst_audio_info_from_caps (&adder->info, caps))
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goto invalid_format;
switch (GST_AUDIO_INFO_FORMAT (&adder->info)) {
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case GST_AUDIO_FORMAT_S8:
adder->func = (GstAdderFunction) add_int8;
break;
case GST_AUDIO_FORMAT_U8:
adder->func = (GstAdderFunction) add_uint8;
break;
case GST_AUDIO_FORMAT_S16:
adder->func = (GstAdderFunction) add_int16;
break;
case GST_AUDIO_FORMAT_U16:
adder->func = (GstAdderFunction) add_uint16;
break;
case GST_AUDIO_FORMAT_S32:
adder->func = (GstAdderFunction) add_int32;
break;
case GST_AUDIO_FORMAT_U32:
adder->func = (GstAdderFunction) add_uint32;
break;
case GST_AUDIO_FORMAT_F32:
adder->func = (GstAdderFunction) add_float32;
break;
case GST_AUDIO_FORMAT_F64:
adder->func = (GstAdderFunction) add_float64;
break;
default:
goto invalid_format;
}
return TRUE;
/* ERRORS */
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invalid_format:
{
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GST_DEBUG_OBJECT (adder, "invalid format set as caps");
return FALSE;
}
}
/* FIXME, the duration query should reflect how long you will produce
* data, that is the amount of stream time until you will emit EOS.
*
* For synchronized mixing this is always the max of all the durations
* of upstream since we emit EOS when all of them finished.
*
* We don't do synchronized mixing so this really depends on where the
* streams where punched in and what their relative offsets are against
* eachother which we can get from the first timestamps we see.
*
* When we add a new stream (or remove a stream) the duration might
* also become invalid again and we need to post a new DURATION
* message to notify this fact to the parent.
* For now we take the max of all the upstream elements so the simple
* cases work at least somewhat.
*/
static gboolean
gst_adder_query_duration (GstAdder * adder, GstQuery * query)
{
gint64 max;
gboolean res;
GstFormat format;
GstIterator *it;
gboolean done;
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GValue item = { 0, };
/* parse format */
gst_query_parse_duration (query, &format, NULL);
max = -1;
res = TRUE;
done = FALSE;
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder));
while (!done) {
GstIteratorResult ires;
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
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GstPad *pad = g_value_get_object (&item);
gint64 duration;
/* ask sink peer for duration */
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res &= gst_pad_peer_query_duration (pad, format, &duration);
/* take max from all valid return values */
if (res) {
/* valid unknown length, stop searching */
if (duration == -1) {
max = duration;
done = TRUE;
}
/* else see if bigger than current max */
else if (duration > max)
max = duration;
}
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g_value_reset (&item);
break;
}
case GST_ITERATOR_RESYNC:
max = -1;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
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g_value_unset (&item);
gst_iterator_free (it);
if (res) {
/* and store the max */
GST_DEBUG_OBJECT (adder, "Total duration in format %s: %"
GST_TIME_FORMAT, gst_format_get_name (format), GST_TIME_ARGS (max));
gst_query_set_duration (query, format, max);
}
return res;
}
static gboolean
gst_adder_query_latency (GstAdder * adder, GstQuery * query)
{
GstClockTime min, max;
gboolean live;
gboolean res;
GstIterator *it;
gboolean done;
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GValue item = { 0, };
res = TRUE;
done = FALSE;
live = FALSE;
min = 0;
max = GST_CLOCK_TIME_NONE;
/* Take maximum of all latency values */
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder));
while (!done) {
GstIteratorResult ires;
ires = gst_iterator_next (it, &item);
switch (ires) {
case GST_ITERATOR_DONE:
done = TRUE;
break;
case GST_ITERATOR_OK:
{
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GstPad *pad = g_value_get_object (&item);
GstQuery *peerquery;
GstClockTime min_cur, max_cur;
gboolean live_cur;
peerquery = gst_query_new_latency ();
/* Ask peer for latency */
res &= gst_pad_peer_query (pad, peerquery);
/* take max from all valid return values */
if (res) {
gst_query_parse_latency (peerquery, &live_cur, &min_cur, &max_cur);
if (min_cur > min)
min = min_cur;
if (max_cur != GST_CLOCK_TIME_NONE &&
((max != GST_CLOCK_TIME_NONE && max_cur > max) ||
(max == GST_CLOCK_TIME_NONE)))
max = max_cur;
live = live || live_cur;
}
gst_query_unref (peerquery);
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g_value_reset (&item);
break;
}
case GST_ITERATOR_RESYNC:
live = FALSE;
min = 0;
max = GST_CLOCK_TIME_NONE;
res = TRUE;
gst_iterator_resync (it);
break;
default:
res = FALSE;
done = TRUE;
break;
}
}
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g_value_unset (&item);
gst_iterator_free (it);
if (res) {
/* store the results */
GST_DEBUG_OBJECT (adder, "Calculated total latency: live %s, min %"
GST_TIME_FORMAT ", max %" GST_TIME_FORMAT,
(live ? "yes" : "no"), GST_TIME_ARGS (min), GST_TIME_ARGS (max));
gst_query_set_latency (query, live, min, max);
}
return res;
}
static gboolean
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gst_adder_src_query (GstPad * pad, GstObject * parent, GstQuery * query)
{
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GstAdder *adder = GST_ADDER (parent);
gboolean res = FALSE;
switch (GST_QUERY_TYPE (query)) {
case GST_QUERY_POSITION:
{
GstFormat format;
gst_query_parse_position (query, &format, NULL);
switch (format) {
case GST_FORMAT_TIME:
/* FIXME, bring to stream time, might be tricky */
gst_query_set_position (query, format, adder->segment.position);
res = TRUE;
break;
case GST_FORMAT_DEFAULT:
gst_query_set_position (query, format, adder->offset);
res = TRUE;
break;
default:
break;
}
break;
}
case GST_QUERY_DURATION:
res = gst_adder_query_duration (adder, query);
break;
case GST_QUERY_LATENCY:
res = gst_adder_query_latency (adder, query);
break;
default:
/* FIXME, needs a custom query handler because we have multiple
* sinkpads */
2011-11-16 16:25:17 +00:00
res = gst_pad_query_default (pad, parent, query);
break;
}
return res;
}
typedef struct
{
GstEvent *event;
gboolean flush;
} EventData;
static gboolean
2011-05-05 14:03:52 +00:00
forward_event_func (const GValue * val, GValue * ret, EventData * data)
{
2011-05-05 14:03:52 +00:00
GstPad *pad = g_value_get_object (val);
GstEvent *event = data->event;
gst_event_ref (event);
GST_LOG_OBJECT (pad, "About to send event %s", GST_EVENT_TYPE_NAME (event));
if (!gst_pad_push_event (pad, event)) {
GST_WARNING_OBJECT (pad, "Sending event %p (%s) failed.",
event, GST_EVENT_TYPE_NAME (event));
/* quick hack to unflush the pads, ideally we need a way to just unflush
* this single collect pad */
if (data->flush)
gst_pad_send_event (pad, gst_event_new_flush_stop (TRUE));
} else {
g_value_set_boolean (ret, TRUE);
GST_LOG_OBJECT (pad, "Sent event %p (%s).",
event, GST_EVENT_TYPE_NAME (event));
}
/* continue on other pads, even if one failed */
return TRUE;
}
/* forwards the event to all sinkpads, takes ownership of the
* event
*
* Returns: TRUE if the event could be forwarded on all
* sinkpads.
*/
static gboolean
forward_event (GstAdder * adder, GstEvent * event, gboolean flush)
{
gboolean ret;
GstIterator *it;
GstIteratorResult ires;
GValue vret = { 0 };
EventData data;
GST_LOG_OBJECT (adder, "Forwarding event %p (%s)", event,
GST_EVENT_TYPE_NAME (event));
data.event = event;
data.flush = flush;
g_value_init (&vret, G_TYPE_BOOLEAN);
g_value_set_boolean (&vret, FALSE);
it = gst_element_iterate_sink_pads (GST_ELEMENT_CAST (adder));
while (TRUE) {
2011-05-05 14:03:52 +00:00
ires =
gst_iterator_fold (it, (GstIteratorFoldFunction) forward_event_func,
&vret, &data);
switch (ires) {
case GST_ITERATOR_RESYNC:
GST_WARNING ("resync");
gst_iterator_resync (it);
g_value_set_boolean (&vret, TRUE);
break;
case GST_ITERATOR_OK:
case GST_ITERATOR_DONE:
ret = g_value_get_boolean (&vret);
goto done;
default:
ret = FALSE;
goto done;
}
}
done:
gst_iterator_free (it);
GST_LOG_OBJECT (adder, "Forwarded event %p (%s), ret=%d", event,
GST_EVENT_TYPE_NAME (event), ret);
gst_event_unref (event);
return ret;
}
static gboolean
2011-11-17 11:48:25 +00:00
gst_adder_src_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstAdder *adder;
gboolean result;
2011-11-17 11:48:25 +00:00
adder = GST_ADDER (parent);
2011-07-25 17:11:59 +00:00
GST_DEBUG_OBJECT (pad, "Got %s event on src pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_SEEK:
{
GstSeekFlags flags;
2011-05-16 11:48:11 +00:00
gdouble rate;
GstSeekType curtype, endtype;
gint64 cur, end;
gboolean flush;
/* parse the seek parameters */
2011-05-16 11:48:11 +00:00
gst_event_parse_seek (event, &rate, NULL, &flags, &curtype,
&cur, &endtype, &end);
if ((curtype != GST_SEEK_TYPE_NONE) && (curtype != GST_SEEK_TYPE_SET)) {
result = FALSE;
GST_DEBUG_OBJECT (adder,
"seeking failed, unhandled seek type for start: %d", curtype);
goto done;
}
if ((endtype != GST_SEEK_TYPE_NONE) && (endtype != GST_SEEK_TYPE_SET)) {
result = FALSE;
GST_DEBUG_OBJECT (adder,
"seeking failed, unhandled seek type for end: %d", endtype);
goto done;
}
flush = (flags & GST_SEEK_FLAG_FLUSH) == GST_SEEK_FLAG_FLUSH;
/* check if we are flushing */
if (flush) {
/* flushing seek, start flush downstream, the flush will be done
* when all pads received a FLUSH_STOP.
* Make sure we accept nothing anymore and return WRONG_STATE.
* We send a flush-start before, to ensure no streaming is done
* as we need to take the stream lock.
*/
gst_pad_push_event (adder->srcpad, gst_event_new_flush_start ());
gst_collect_pads2_set_flushing (adder->collect, TRUE);
/* We can't send FLUSH_STOP here since upstream could start pushing data
* after we unlock adder->collect.
* We set flush_stop_pending to TRUE instead and send FLUSH_STOP after
* forwarding the seek upstream or from gst_adder_collected,
* whichever happens first.
*/
g_atomic_int_set (&adder->flush_stop_pending, TRUE);
}
GST_DEBUG_OBJECT (adder, "handling seek event: %" GST_PTR_FORMAT, event);
/* now wait for the collected to be finished and mark a new
* segment. After we have the lock, no collect function is running and no
* new collect function will be called for as long as we're flushing. */
GST_COLLECT_PADS2_STREAM_LOCK (adder->collect);
2011-05-16 11:48:11 +00:00
adder->segment.rate = rate;
if (curtype == GST_SEEK_TYPE_SET)
2011-05-16 11:48:11 +00:00
adder->segment.start = cur;
else
2011-05-16 11:48:11 +00:00
adder->segment.start = 0;
if (endtype == GST_SEEK_TYPE_SET)
2011-05-16 11:48:11 +00:00
adder->segment.stop = end;
else
2011-05-16 11:48:11 +00:00
adder->segment.stop = GST_CLOCK_TIME_NONE;
if (flush) {
/* Yes, we need to call _set_flushing again *WHEN* the streaming threads
* have stopped so that the cookie gets properly updated. */
2011-12-14 18:26:07 +00:00
gst_collect_pads2_set_flushing (adder->collect, TRUE);
}
GST_COLLECT_PADS2_STREAM_UNLOCK (adder->collect);
GST_DEBUG_OBJECT (adder, "forwarding seek event: %" GST_PTR_FORMAT,
event);
/* we're forwarding seek to all upstream peers and wait for one to reply
* with a newsegment-event before we send a newsegment-event downstream */
g_atomic_int_set (&adder->wait_for_new_segment, TRUE);
result = forward_event (adder, event, flush);
if (!result) {
/* seek failed. maybe source is a live source. */
GST_DEBUG_OBJECT (adder, "seeking failed");
}
if (g_atomic_int_compare_and_exchange (&adder->flush_stop_pending,
TRUE, FALSE)) {
GST_DEBUG_OBJECT (adder, "pending flush stop");
gst_pad_push_event (adder->srcpad, gst_event_new_flush_stop (TRUE));
}
break;
}
case GST_EVENT_QOS:
/* QoS might be tricky */
result = FALSE;
break;
case GST_EVENT_NAVIGATION:
/* navigation is rather pointless. */
result = FALSE;
break;
default:
/* just forward the rest for now */
GST_DEBUG_OBJECT (adder, "forward unhandled event: %s",
GST_EVENT_TYPE_NAME (event));
result = forward_event (adder, event, FALSE);
break;
}
done:
return result;
}
static gboolean
2011-11-17 11:48:25 +00:00
gst_adder_sink_event (GstPad * pad, GstObject * parent, GstEvent * event)
{
GstAdder *adder;
gboolean ret = TRUE;
2011-11-17 11:48:25 +00:00
adder = GST_ADDER (parent);
2011-07-25 17:11:59 +00:00
GST_DEBUG_OBJECT (pad, "Got %s event on sink pad",
GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_CAPS:
{
GstCaps *caps;
gst_event_parse_caps (event, &caps);
ret = gst_adder_setcaps (adder, pad, caps);
gst_event_unref (event);
goto beach;
}
case GST_EVENT_FLUSH_STOP:
/* we received a flush-stop. The collect_event function will call the
* gst_adder_event function we have set on the GstCollectPads2, so we
* have control over whether the event is sent past our element.
* We will only forward it when flush_stop_pending is set, and we will
* unset it then.
*/
GST_COLLECT_PADS2_STREAM_LOCK (adder->collect);
g_atomic_int_set (&adder->new_segment_pending, TRUE);
/* Clear pending tags */
if (adder->pending_events) {
g_list_foreach (adder->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (adder->pending_events);
adder->pending_events = NULL;
}
GST_COLLECT_PADS2_STREAM_UNLOCK (adder->collect);
break;
case GST_EVENT_TAG:
GST_COLLECT_PADS2_STREAM_LOCK (adder->collect);
2009-12-24 12:58:52 +00:00
/* collect tags here so we can push them out when we collect data */
adder->pending_events = g_list_append (adder->pending_events, event);
GST_COLLECT_PADS2_STREAM_UNLOCK (adder->collect);
goto beach;
2011-08-17 08:47:20 +00:00
case GST_EVENT_SEGMENT:
if (g_atomic_int_compare_and_exchange (&adder->wait_for_new_segment,
TRUE, FALSE)) {
/* make sure we push a new segment, to inform about new basetime
* see FIXME in gst_adder_collected() */
g_atomic_int_set (&adder->new_segment_pending, TRUE);
}
break;
default:
break;
}
2011-12-14 18:26:07 +00:00
/* now GstCollectPads2 can take care of the rest, e.g. EOS */
2011-11-17 11:48:25 +00:00
ret = adder->collect_event (pad, parent, event);
beach:
return ret;
}
2010-09-01 07:06:09 +00:00
static void
gst_adder_class_init (GstAdderClass * klass)
{
GObjectClass *gobject_class = (GObjectClass *) klass;
GstElementClass *gstelement_class = (GstElementClass *) klass;
2010-09-01 07:06:09 +00:00
gobject_class->set_property = gst_adder_set_property;
gobject_class->get_property = gst_adder_get_property;
gobject_class->dispose = gst_adder_dispose;
2009-07-10 22:24:36 +00:00
/**
* GstAdder:caps:
*
* Since: 0.10.24
*/
g_object_class_install_property (gobject_class, PROP_FILTER_CAPS,
2009-07-10 22:24:36 +00:00
g_param_spec_boxed ("caps", "Target caps",
"Set target format for mixing (NULL means ANY). "
"Setting this property takes a reference to the supplied GstCaps "
"object.", GST_TYPE_CAPS,
G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_adder_src_template));
gst_element_class_add_pad_template (gstelement_class,
gst_static_pad_template_get (&gst_adder_sink_template));
gst_element_class_set_details_simple (gstelement_class, "Adder",
"Generic/Audio",
"Add N audio channels together",
"Thomas Vander Stichele <thomas at apestaart dot org>");
gstelement_class->request_new_pad =
GST_DEBUG_FUNCPTR (gst_adder_request_new_pad);
gstelement_class->release_pad = GST_DEBUG_FUNCPTR (gst_adder_release_pad);
gstelement_class->change_state = GST_DEBUG_FUNCPTR (gst_adder_change_state);
}
static void
gst_adder_init (GstAdder * adder)
{
GstPadTemplate *template;
template = gst_static_pad_template_get (&gst_adder_src_template);
adder->srcpad = gst_pad_new_from_template (template, "src");
gst_object_unref (template);
gst_pad_set_query_function (adder->srcpad,
GST_DEBUG_FUNCPTR (gst_adder_src_query));
gst_pad_set_event_function (adder->srcpad,
GST_DEBUG_FUNCPTR (gst_adder_src_event));
GST_PAD_SET_PROXY_CAPS (adder->srcpad);
gst_element_add_pad (GST_ELEMENT (adder), adder->srcpad);
2011-08-19 15:05:55 +00:00
gst_audio_info_init (&adder->info);
adder->padcount = 0;
adder->func = NULL;
adder->filter_caps = NULL;
/* keep track of the sinkpads requested */
2011-12-14 18:26:07 +00:00
adder->collect = gst_collect_pads2_new ();
gst_collect_pads2_set_function (adder->collect,
GST_DEBUG_FUNCPTR (gst_adder_collected), adder);
2011-12-14 18:26:07 +00:00
gst_collect_pads2_set_clip_function (adder->collect,
GST_DEBUG_FUNCPTR (gst_adder_do_clip), adder);
gst_collect_pads2_set_event_function (adder->collect,
GST_DEBUG_FUNCPTR (gst_adder_event), adder);
}
static void
gst_adder_dispose (GObject * object)
{
GstAdder *adder = GST_ADDER (object);
if (adder->collect) {
gst_object_unref (adder->collect);
adder->collect = NULL;
}
gst_caps_replace (&adder->filter_caps, NULL);
if (adder->pending_events) {
g_list_foreach (adder->pending_events, (GFunc) gst_event_unref, NULL);
g_list_free (adder->pending_events);
adder->pending_events = NULL;
}
G_OBJECT_CLASS (parent_class)->dispose (object);
}
static void
gst_adder_set_property (GObject * object, guint prop_id,
const GValue * value, GParamSpec * pspec)
{
GstAdder *adder = GST_ADDER (object);
switch (prop_id) {
case PROP_FILTER_CAPS:{
GstCaps *new_caps = NULL;
GstCaps *old_caps;
const GstCaps *new_caps_val = gst_value_get_caps (value);
if (new_caps_val != NULL) {
new_caps = (GstCaps *) new_caps_val;
gst_caps_ref (new_caps);
}
GST_OBJECT_LOCK (adder);
old_caps = adder->filter_caps;
adder->filter_caps = new_caps;
GST_OBJECT_UNLOCK (adder);
if (old_caps)
gst_caps_unref (old_caps);
GST_DEBUG_OBJECT (adder, "set new caps %" GST_PTR_FORMAT, new_caps);
break;
}
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_adder_get_property (GObject * object, guint prop_id, GValue * value,
GParamSpec * pspec)
{
GstAdder *adder = GST_ADDER (object);
switch (prop_id) {
case PROP_FILTER_CAPS:
GST_OBJECT_LOCK (adder);
gst_value_set_caps (value, adder->filter_caps);
GST_OBJECT_UNLOCK (adder);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static GstPad *
gst_adder_request_new_pad (GstElement * element, GstPadTemplate * templ,
2011-05-10 14:44:37 +00:00
const gchar * unused, const GstCaps * caps)
{
gchar *name;
GstAdder *adder;
GstPad *newpad;
gint padcount;
if (templ->direction != GST_PAD_SINK)
goto not_sink;
adder = GST_ADDER (element);
/* increment pad counter */
padcount = g_atomic_int_add (&adder->padcount, 1);
name = g_strdup_printf ("sink_%u", padcount);
newpad = gst_pad_new_from_template (templ, name);
GST_DEBUG_OBJECT (adder, "request new pad %s", name);
g_free (name);
gst_pad_set_query_function (newpad, GST_DEBUG_FUNCPTR (gst_adder_sink_query));
2011-12-14 18:26:07 +00:00
gst_collect_pads2_add_pad (adder->collect, newpad, sizeof (GstCollectData2));
/* FIXME: hacked way to override/extend the event function of
2011-12-14 18:26:07 +00:00
* GstCollectPads2; because it sets its own event function giving the
* element no access to events */
adder->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (newpad);
gst_pad_set_event_function (newpad, GST_DEBUG_FUNCPTR (gst_adder_sink_event));
/* takes ownership of the pad */
if (!gst_element_add_pad (GST_ELEMENT (adder), newpad))
goto could_not_add;
return newpad;
/* errors */
not_sink:
{
g_warning ("gstadder: request new pad that is not a SINK pad\n");
return NULL;
}
could_not_add:
{
GST_DEBUG_OBJECT (adder, "could not add pad");
2011-12-14 18:26:07 +00:00
gst_collect_pads2_remove_pad (adder->collect, newpad);
gst_object_unref (newpad);
return NULL;
}
}
static void
gst_adder_release_pad (GstElement * element, GstPad * pad)
{
GstAdder *adder;
adder = GST_ADDER (element);
GST_DEBUG_OBJECT (adder, "release pad %s:%s", GST_DEBUG_PAD_NAME (pad));
if (adder->collect)
gst_collect_pads2_remove_pad (adder->collect, pad);
gst_element_remove_pad (element, pad);
}
2011-12-14 18:26:07 +00:00
static GstFlowReturn
gst_adder_do_clip (GstCollectPads2 * pads, GstCollectData2 * data,
GstBuffer * buffer, GstBuffer ** out, gpointer user_data)
{
GstAdder *adder = GST_ADDER (user_data);
2011-08-19 15:05:55 +00:00
gint rate, bpf;
rate = GST_AUDIO_INFO_RATE (&adder->info);
bpf = GST_AUDIO_INFO_BPF (&adder->info);
2011-08-19 15:05:55 +00:00
buffer = gst_audio_buffer_clip (buffer, &data->segment, rate, bpf);
2011-12-14 18:26:07 +00:00
*out = buffer;
return GST_FLOW_OK;
}
static GstFlowReturn
2011-12-14 18:26:07 +00:00
gst_adder_collected (GstCollectPads2 * pads, gpointer user_data)
{
/*
* combine streams by adding data values
* basic algorithm :
* - this function is called when all pads have a buffer
* - get available bytes on all pads.
* - repeat for each input pad :
* - read available bytes, copy or add to target buffer
* - if there's an EOS event, remove the input channel
* - push out the output buffer
*
* todo:
* - would be nice to have a mixing mode, where instead of adding we mix
* - for float we could downscale after collect loop
* - for int we need to downscale each input to avoid clipping or
* mix into a temp (float) buffer and scale afterwards as well
*/
GstAdder *adder;
GSList *collected, *next = NULL;
GstFlowReturn ret;
GstBuffer *outbuf = NULL, *gapbuf = NULL;
2012-01-20 15:11:54 +00:00
GstMapInfo outmap = { NULL };
guint outsize;
gint64 next_offset;
gint64 next_timestamp;
2011-08-19 15:05:55 +00:00
gint rate, bps, bpf;
adder = GST_ADDER (user_data);
/* this is fatal */
if (G_UNLIKELY (adder->func == NULL))
goto not_negotiated;
if (g_atomic_int_compare_and_exchange (&adder->flush_stop_pending,
TRUE, FALSE)) {
GST_DEBUG_OBJECT (adder, "pending flush stop");
gst_pad_push_event (adder->srcpad, gst_event_new_flush_stop (TRUE));
}
/* get available bytes for reading, this can be 0 which could mean empty
* buffers or EOS, which we will catch when we loop over the pads. */
2011-12-14 18:26:07 +00:00
outsize = gst_collect_pads2_available (pads);
/* can only happen when no pads to collect or all EOS */
if (outsize == 0)
goto eos;
2011-08-19 15:05:55 +00:00
rate = GST_AUDIO_INFO_RATE (&adder->info);
bps = GST_AUDIO_INFO_BPS (&adder->info);
bpf = GST_AUDIO_INFO_BPF (&adder->info);
GST_LOG_OBJECT (adder,
"starting to cycle through channels, %d bytes available (bps = %d, bpf = %d)",
outsize, bps, bpf);
for (collected = pads->data; collected; collected = next) {
2011-12-14 18:26:07 +00:00
GstCollectData2 *collect_data;
GstBuffer *inbuf;
gboolean is_gap;
/* take next to see if this is the last collectdata */
next = g_slist_next (collected);
2011-12-14 18:26:07 +00:00
collect_data = (GstCollectData2 *) collected->data;
/* get a buffer of size bytes, if we get a buffer, it is at least outsize
* bytes big. */
2011-12-14 18:26:07 +00:00
inbuf = gst_collect_pads2_take_buffer (pads, collect_data, outsize);
/* NULL means EOS or an empty buffer so we still need to flush in
* case of an empty buffer. */
if (inbuf == NULL) {
GST_LOG_OBJECT (adder, "channel %p: no bytes available", collect_data);
continue;
}
is_gap = GST_BUFFER_FLAG_IS_SET (inbuf, GST_BUFFER_FLAG_GAP);
/* Try to make an output buffer */
if (outbuf == NULL) {
/* if this is a gap buffer but we have some more pads to check, skip it.
* If we are at the last buffer, take it, regardless if it is a GAP
* buffer or not. */
if (is_gap && next) {
GST_DEBUG_OBJECT (adder, "skipping, non-last GAP buffer");
/* we keep the GAP buffer, if we don't have anymore buffers (all pads
* EOS, we can use this one as the output buffer. */
if (gapbuf == NULL)
gapbuf = inbuf;
else
gst_buffer_unref (inbuf);
continue;
}
GST_LOG_OBJECT (adder, "channel %p: preparing output buffer of %d bytes",
collect_data, outsize);
/* make data and metadata writable, can simply return the inbuf when we
* are the only one referencing this buffer. If this is the last (and
* only) GAP buffer, it will automatically copy the GAP flag. */
outbuf = gst_buffer_make_writable (inbuf);
2012-01-20 15:11:54 +00:00
gst_buffer_map (outbuf, &outmap, GST_MAP_WRITE);
} else {
if (!is_gap) {
/* we had a previous output buffer, mix this non-GAP buffer */
2012-01-20 15:11:54 +00:00
GstMapInfo inmap;
2012-01-20 15:11:54 +00:00
gst_buffer_map (inbuf, &inmap, GST_MAP_READ);
/* all buffers should have outsize, there are no short buffers because we
* asked for the max size above */
2012-01-20 15:11:54 +00:00
g_assert (inmap.size == outmap.size);
GST_LOG_OBJECT (adder, "channel %p: mixing %" G_GSIZE_FORMAT " bytes"
2012-01-20 15:11:54 +00:00
" from data %p", collect_data, inmap.size, inmap.data);
/* further buffers, need to add them */
2012-01-20 15:11:54 +00:00
adder->func ((gpointer) outmap.data, (gpointer) inmap.data,
inmap.size / bps);
gst_buffer_unmap (inbuf, &inmap);
} else {
/* skip gap buffer */
GST_LOG_OBJECT (adder, "channel %p: skipping GAP buffer", collect_data);
}
gst_buffer_unref (inbuf);
}
}
if (outbuf)
2012-01-20 15:11:54 +00:00
gst_buffer_unmap (outbuf, &outmap);
if (outbuf == NULL) {
/* no output buffer, reuse one of the GAP buffers then if we have one */
if (gapbuf) {
GST_LOG_OBJECT (adder, "reusing GAP buffer %p", gapbuf);
outbuf = gapbuf;
} else
/* assume EOS otherwise, this should not happen, really */
goto eos;
} else if (gapbuf)
/* we had an output buffer, unref the gapbuffer we kept */
gst_buffer_unref (gapbuf);
if (g_atomic_int_compare_and_exchange (&adder->new_segment_pending, TRUE,
FALSE)) {
GstEvent *event;
/* FIXME, use rate/applied_rate as set on all sinkpads.
* - currently we just set rate as received from last seek-event
*
* When seeking we set the start and stop positions as given in the seek
* event. We also adjust offset & timestamp accordingly.
* This basically ignores all newsegments sent by upstream.
*/
2011-05-16 11:48:11 +00:00
event = gst_event_new_segment (&adder->segment);
if (adder->segment.rate > 0.0) {
adder->segment.position = adder->segment.start;
} else {
2011-05-16 11:48:11 +00:00
adder->segment.position = adder->segment.stop;
}
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adder->offset = gst_util_uint64_scale (adder->segment.position,
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rate, GST_SECOND);
GST_INFO_OBJECT (adder, "seg_start %" G_GUINT64_FORMAT ", seg_end %"
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G_GUINT64_FORMAT, adder->segment.start, adder->segment.stop);
GST_INFO_OBJECT (adder, "timestamp %" G_GINT64_FORMAT ",new offset %"
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G_GINT64_FORMAT, adder->segment.position, adder->offset);
if (event) {
if (!gst_pad_push_event (adder->srcpad, event)) {
GST_WARNING_OBJECT (adder->srcpad, "Sending event failed");
}
} else {
GST_WARNING_OBJECT (adder->srcpad, "Creating new segment event for "
"start:%" G_GINT64_FORMAT " end:%" G_GINT64_FORMAT " failed",
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adder->segment.start, adder->segment.stop);
}
}
if (G_UNLIKELY (adder->pending_events)) {
GList *tmp = adder->pending_events;
while (tmp) {
GstEvent *ev = (GstEvent *) tmp->data;
gst_pad_push_event (adder->srcpad, ev);
tmp = g_list_next (tmp);
}
g_list_free (adder->pending_events);
adder->pending_events = NULL;
}
/* for the next timestamp, use the sample counter, which will
* never accumulate rounding errors */
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if (adder->segment.rate > 0.0) {
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next_offset = adder->offset + outsize / bpf;
} else {
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next_offset = adder->offset - outsize / bpf;
}
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next_timestamp = gst_util_uint64_scale (next_offset, GST_SECOND, rate);
/* set timestamps on the output buffer */
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if (adder->segment.rate > 0.0) {
GST_BUFFER_TIMESTAMP (outbuf) = adder->segment.position;
GST_BUFFER_OFFSET (outbuf) = adder->offset;
GST_BUFFER_OFFSET_END (outbuf) = next_offset;
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GST_BUFFER_DURATION (outbuf) = next_timestamp - adder->segment.position;
} else {
GST_BUFFER_TIMESTAMP (outbuf) = next_timestamp;
GST_BUFFER_OFFSET (outbuf) = next_offset;
GST_BUFFER_OFFSET_END (outbuf) = adder->offset;
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GST_BUFFER_DURATION (outbuf) = adder->segment.position - next_timestamp;
}
adder->offset = next_offset;
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adder->segment.position = next_timestamp;
/* send it out */
GST_LOG_OBJECT (adder, "pushing outbuf %p, timestamp %" GST_TIME_FORMAT
" offset %" G_GINT64_FORMAT, outbuf,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_BUFFER_OFFSET (outbuf));
ret = gst_pad_push (adder->srcpad, outbuf);
GST_LOG_OBJECT (adder, "pushed outbuf, result = %s", gst_flow_get_name (ret));
return ret;
/* ERRORS */
not_negotiated:
{
GST_ELEMENT_ERROR (adder, STREAM, FORMAT, (NULL),
("Unknown data received, not negotiated"));
return GST_FLOW_NOT_NEGOTIATED;
}
eos:
{
GST_DEBUG_OBJECT (adder, "no data available, must be EOS");
gst_pad_push_event (adder->srcpad, gst_event_new_eos ());
return GST_FLOW_EOS;
}
}
static gboolean
gst_adder_event (GstCollectPads2 * pads, GstCollectData2 * pad,
GstEvent * event, gpointer user_data)
{
GstAdder *adder = GST_ADDER (user_data);
if (GST_EVENT_TYPE (event) == GST_EVENT_FLUSH_STOP) {
if (g_atomic_int_compare_and_exchange (&adder->flush_stop_pending,
TRUE, FALSE)) {
return gst_pad_event_default (pad->pad, GST_OBJECT (user_data), event);
} else {
gst_event_unref (event);
return TRUE;
}
} else {
if (GST_EVENT_TYPE (event) == GST_EVENT_SEGMENT ||
GST_EVENT_TYPE (event) == GST_EVENT_CAPS ||
GST_EVENT_TYPE (event) == GST_EVENT_EOS) {
gst_event_unref (event);
return TRUE;
} else {
return gst_pad_event_default (pad->pad, GST_OBJECT (user_data), event);
}
}
}
static GstStateChangeReturn
gst_adder_change_state (GstElement * element, GstStateChange transition)
{
GstAdder *adder;
GstStateChangeReturn ret;
adder = GST_ADDER (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
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adder->segment.position = 0;
adder->offset = 0;
adder->flush_stop_pending = FALSE;
adder->new_segment_pending = TRUE;
adder->wait_for_new_segment = FALSE;
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gst_segment_init (&adder->segment, GST_FORMAT_TIME);
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gst_collect_pads2_start (adder->collect);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
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/* need to unblock the collectpads2 before calling the
* parent change_state so that streaming can finish */
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gst_collect_pads2_stop (adder->collect);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
default:
break;
}
return ret;
}
static gboolean
plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "adder", 0,
"audio channel mixing element");
gst_adder_orc_init ();
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if (!gst_element_register (plugin, "adder", GST_RANK_NONE, GST_TYPE_ADDER)) {
return FALSE;
}
return TRUE;
}
GST_PLUGIN_DEFINE (GST_VERSION_MAJOR,
GST_VERSION_MINOR,
"adder",
"Adds multiple streams",
plugin_init, VERSION, "LGPL", GST_PACKAGE_NAME, GST_PACKAGE_ORIGIN)