gstreamer/ext/ogg/gstoggmux.c

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/* OGG muxer plugin for GStreamer
* Copyright (C) 2004 Wim Taymans <wim@fluendo.com>
* Copyright (C) 2006 Thomas Vander Stichele <thomas at apestaart dot 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.
*/
docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux * 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/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: Update introspection data. * ext/ogg/gstoggmux.c: Document oggmux. * gst/playback/gstdecodebin2.c: Don't use gtk-doc style comment start for private stuff, but make it formatted like this for consistency.
2008-04-03 14:58:06 +00:00
/**
* SECTION:element-oggmux
2010-03-29 09:53:11 +00:00
* @see_also: <link linkend="gst-plugins-base-plugins-oggdemux">oggdemux</link>
docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux * 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/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: Update introspection data. * ext/ogg/gstoggmux.c: Document oggmux. * gst/playback/gstdecodebin2.c: Don't use gtk-doc style comment start for private stuff, but make it formatted like this for consistency.
2008-04-03 14:58:06 +00:00
*
* This element merges streams (audio and video) into ogg files.
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
*
* <refsect2>
docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux * 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/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: Update introspection data. * ext/ogg/gstoggmux.c: Document oggmux. * gst/playback/gstdecodebin2.c: Don't use gtk-doc style comment start for private stuff, but make it formatted like this for consistency.
2008-04-03 14:58:06 +00:00
* <title>Example pipelines</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
* |[
* gst-launch v4l2src num-buffers=500 ! video/x-raw-yuv,width=320,height=240 ! ffmpegcolorspace ! theoraenc ! oggmux ! filesink location=video.ogg
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
* ]| Encodes a video stream captured from a v4l2-compatible camera to Ogg/Theora
* (the encoding will stop automatically after 500 frames)
docs/plugins/: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-overrides.txt: * docs/plugins/gst-plugins-base-plugins-sections.txt: Add playbin, playbin2, decodebin, decodebin2, uridecodebin and oggmux * 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/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: Update introspection data. * ext/ogg/gstoggmux.c: Document oggmux. * gst/playback/gstdecodebin2.c: Don't use gtk-doc style comment start for private stuff, but make it formatted like this for consistency.
2008-04-03 14:58:06 +00:00
* </refsect2>
*
* Last reviewed on 2008-02-06 (0.10.17)
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include <gst/gst.h>
#include <gst/base/gstcollectpads.h>
2010-05-05 11:59:57 +00:00
#include <gst/tag/tag.h>
docs/plugins/: First round of plugin docs cleansups. 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: * docs/plugins/gst-plugins-base-plugins.hierarchy: * docs/plugins/gst-plugins-base-plugins.interfaces: * docs/plugins/gst-plugins-base-plugins.prerequisites: First round of plugin docs cleansups. * 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-subparse.xml: * docs/plugins/inspect/plugin-tcp.xml: * docs/plugins/inspect/plugin-theora.xml: * docs/plugins/inspect/plugin-typefindfunctions.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: Regenerate. * ext/ogg/Makefile.am: * ext/ogg/gstoggmux.c: * ext/ogg/gstoggmux.h: Add header for oggmux. the c-file needs a doc blob still.
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#include "gstoggmux.h"
/* memcpy - if someone knows a way to get rid of it, please speak up
* note: the ogg docs even say you need this... */
#include <string.h>
#include <time.h>
#include <stdlib.h> /* rand, srand, atoi */
GST_DEBUG_CATEGORY_STATIC (gst_ogg_mux_debug);
#define GST_CAT_DEFAULT gst_ogg_mux_debug
/* This isn't generally what you'd want with an end-time macro, because
technically the end time of a buffer with invalid duration is invalid. But
for sorting ogg pages this is what we want. */
#define GST_BUFFER_END_TIME(buf) \
(GST_BUFFER_DURATION_IS_VALID (buf) \
? GST_BUFFER_TIMESTAMP (buf) + GST_BUFFER_DURATION (buf) \
: GST_BUFFER_TIMESTAMP (buf))
#define GST_GP_FORMAT "[gp %8" G_GINT64_FORMAT "]"
#define GST_GP_CAST(_gp) ((gint64) _gp)
typedef enum
{
GST_OGG_FLAG_BOS = GST_ELEMENT_FLAG_LAST,
GST_OGG_FLAG_EOS
}
GstOggFlag;
/* OggMux signals and args */
enum
{
/* FILL ME */
LAST_SIGNAL
};
/* set to 0.5 seconds by default */
#define DEFAULT_MAX_DELAY G_GINT64_CONSTANT(500000000)
#define DEFAULT_MAX_PAGE_DELAY G_GINT64_CONSTANT(500000000)
#define DEFAULT_MAX_TOLERANCE G_GINT64_CONSTANT(40000000)
enum
{
ARG_0,
ARG_MAX_DELAY,
ARG_MAX_PAGE_DELAY,
ARG_MAX_TOLERANCE
};
static GstStaticPadTemplate src_factory = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS ("application/ogg")
);
static GstStaticPadTemplate sink_factory = GST_STATIC_PAD_TEMPLATE ("sink_%d",
GST_PAD_SINK,
GST_PAD_REQUEST,
GST_STATIC_CAPS ("video/x-theora; "
"audio/x-vorbis; audio/x-flac; audio/x-speex; audio/x-celt; "
"application/x-ogm-video; application/x-ogm-audio; video/x-dirac; "
2010-05-05 11:59:57 +00:00
"video/x-smoke; video/x-vp8; text/x-cmml, encoded = (boolean) TRUE; "
"subtitle/x-kate; application/x-kate")
);
static void gst_ogg_mux_base_init (gpointer g_class);
static void gst_ogg_mux_class_init (GstOggMuxClass * klass);
static void gst_ogg_mux_init (GstOggMux * ogg_mux);
static void gst_ogg_mux_finalize (GObject * object);
static GstFlowReturn
gst_ogg_mux_collected (GstCollectPads * pads, GstOggMux * ogg_mux);
static gboolean gst_ogg_mux_handle_src_event (GstPad * pad, GstEvent * event);
static GstPad *gst_ogg_mux_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * name);
static void gst_ogg_mux_release_pad (GstElement * element, GstPad * pad);
static void gst_ogg_mux_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec);
static void gst_ogg_mux_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec);
static GstStateChangeReturn gst_ogg_mux_change_state (GstElement * element,
GstStateChange transition);
static GstElementClass *parent_class = NULL;
/*static guint gst_ogg_mux_signals[LAST_SIGNAL] = { 0 }; */
GType
gst_ogg_mux_get_type (void)
{
static GType ogg_mux_type = 0;
if (G_UNLIKELY (ogg_mux_type == 0)) {
static const GTypeInfo ogg_mux_info = {
sizeof (GstOggMuxClass),
gst_ogg_mux_base_init,
NULL,
(GClassInitFunc) gst_ogg_mux_class_init,
NULL,
NULL,
sizeof (GstOggMux),
0,
(GInstanceInitFunc) gst_ogg_mux_init,
};
2009-05-14 09:41:13 +00:00
static const GInterfaceInfo preset_info = {
NULL,
NULL,
NULL
};
ogg_mux_type =
g_type_register_static (GST_TYPE_ELEMENT, "GstOggMux", &ogg_mux_info,
0);
2009-05-14 09:41:13 +00:00
g_type_add_interface_static (ogg_mux_type, GST_TYPE_PRESET, &preset_info);
}
return ogg_mux_type;
}
static void
gst_ogg_mux_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_simple (element_class,
"Ogg muxer", "Codec/Muxer",
"mux ogg streams (info about ogg: http://xiph.org)",
"Wim Taymans <wim@fluendo.com>");
}
static void
gst_ogg_mux_class_init (GstOggMuxClass * klass)
{
GObjectClass *gobject_class;
GstElementClass *gstelement_class;
gobject_class = (GObjectClass *) klass;
gstelement_class = (GstElementClass *) klass;
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent) Original commit message from CVS: * ext/alsa/gstalsamixeroptions.c: (gst_alsa_mixer_options_class_init): * ext/alsa/gstalsamixertrack.c: (gst_alsa_mixer_track_class_init): * ext/ogg/gstoggdemux.c: (gst_ogg_pad_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/ogg/gstoggparse.c: (gst_ogg_parse_class_init): * gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_class_init): * gst-libs/gst/audio/gstaudiofilter.c: (gst_audio_filter_class_init): * gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_class_init): * gst-libs/gst/audio/gstaudiosrc.c: (gst_audioringbuffer_class_init): * gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_class_init): * gst-libs/gst/interfaces/colorbalancechannel.c: (gst_color_balance_channel_class_init): * gst-libs/gst/interfaces/mixeroptions.c: (gst_mixer_options_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/interfaces/tunerchannel.c: (gst_tuner_channel_class_init): * gst-libs/gst/interfaces/tunernorm.c: (gst_tuner_norm_class_init): * gst-libs/gst/netbuffer/gstnetbuffer.c: (gst_netbuffer_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_stream_selector_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * sys/v4l/gstv4lcolorbalance.c: (gst_v4l_color_balance_channel_class_init): * sys/v4l/gstv4ljpegsrc.c: (gst_v4ljpegsrc_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4ltuner.c: (gst_v4l_tuner_channel_class_init), (gst_v4l_tuner_norm_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): * tests/old/testsuite/alsa/sinesrc.c: (sinesrc_class_init): Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
2006-04-08 21:02:53 +00:00
parent_class = g_type_class_peek_parent (klass);
gobject_class->finalize = gst_ogg_mux_finalize;
gobject_class->get_property = gst_ogg_mux_get_property;
gobject_class->set_property = gst_ogg_mux_set_property;
gstelement_class->request_new_pad = gst_ogg_mux_request_new_pad;
gstelement_class->release_pad = gst_ogg_mux_release_pad;
g_object_class_install_property (gobject_class, ARG_MAX_DELAY,
g_param_spec_uint64 ("max-delay", "Max delay",
"Maximum delay in multiplexing streams", 0, G_MAXUINT64,
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
DEFAULT_MAX_DELAY,
(GParamFlags) G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_MAX_PAGE_DELAY,
g_param_spec_uint64 ("max-page-delay", "Max page delay",
"Maximum delay for sending out a page", 0, G_MAXUINT64,
Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory u... Original commit message from CVS: * configure.ac: * ext/alsa/gstalsamixerelement.c: (gst_alsa_mixer_element_class_init): * ext/alsa/gstalsasink.c: (gst_alsasink_class_init): * ext/alsa/gstalsasrc.c: (gst_alsasrc_class_init): * ext/cdparanoia/gstcdparanoiasrc.c: (gst_cd_paranoia_src_class_init): * ext/gio/gstgiosink.c: (gst_gio_sink_class_init): * ext/gio/gstgiosrc.c: (gst_gio_src_class_init): * ext/gio/gstgiostreamsink.c: (gst_gio_stream_sink_class_init): * ext/gio/gstgiostreamsrc.c: (gst_gio_stream_src_class_init): * ext/gnomevfs/gstgnomevfssink.c: (gst_gnome_vfs_sink_class_init): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_class_init): * ext/ogg/gstoggmux.c: (gst_ogg_mux_class_init): * ext/pango/gsttextoverlay.c: (gst_text_overlay_class_init): * ext/pango/gsttextrender.c: (gst_text_render_class_init): * ext/theora/theoradec.c: (gst_theora_dec_class_init): * ext/theora/theoraenc.c: (gst_theora_enc_class_init): * ext/theora/theoraparse.c: (gst_theora_parse_class_init): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_class_init): * gst-libs/gst/audio/gstaudiofiltertemplate.c: (gst_audio_filter_template_class_init): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init): * gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_class_init): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_class_init): * gst-libs/gst/interfaces/mixertrack.c: (gst_mixer_track_class_init): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_class_init): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_class_init): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_class_init): * gst/audiorate/gstaudiorate.c: (gst_audio_rate_class_init): * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/audiotestsrc/gstaudiotestsrc.c: (gst_audio_test_src_class_init): * gst/gdp/gstgdppay.c: (gst_gdp_pay_class_init): * gst/playback/gstdecodebin2.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init), (preroll_unlinked): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/playback/gstplaybin2.c: (gst_play_bin_class_init): * gst/playback/gstplaysink.c: (gst_play_sink_class_init): * gst/playback/gstqueue2.c: (gst_queue_class_init): * gst/playback/gststreaminfo.c: (gst_stream_info_class_init): * gst/playback/gststreamselector.c: (gst_selector_pad_class_init), (gst_stream_selector_class_init): * gst/playback/gsturidecodebin.c: (gst_uri_decode_bin_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videorate/gstvideorate.c: (gst_video_rate_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): * sys/v4l/gstv4lelement.c: (gst_v4lelement_class_init): * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_class_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_class_init): * sys/v4l/gstv4lsrc.c: (gst_v4lsrc_class_init): * sys/ximage/ximagesink.c: (gst_ximagesink_class_init): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_class_init): Use G_PARAM_STATIC_STRINGS everywhere for GParamSpecs that use static strings (i.e. all). This gives us less memory usage, fewer allocations and thus less memory defragmentation. Depend on core CVS for this. Fixes bug #523806.
2008-03-22 15:00:53 +00:00
DEFAULT_MAX_PAGE_DELAY,
(GParamFlags) G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
g_object_class_install_property (gobject_class, ARG_MAX_TOLERANCE,
g_param_spec_uint64 ("max-tolerance", "Max time tolerance",
"Maximum timestamp difference for maintaining perfect granules",
0, G_MAXUINT64, DEFAULT_MAX_TOLERANCE,
(GParamFlags) G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
gstelement_class->change_state = gst_ogg_mux_change_state;
}
#if 0
static const GstEventMask *
gst_ogg_mux_get_sink_event_masks (GstPad * pad)
{
static const GstEventMask gst_ogg_mux_sink_event_masks[] = {
{GST_EVENT_EOS, 0},
{GST_EVENT_DISCONTINUOUS, 0},
{0,}
};
return gst_ogg_mux_sink_event_masks;
}
#endif
static void
gst_ogg_mux_clear (GstOggMux * ogg_mux)
{
ogg_mux->pulling = NULL;
ogg_mux->need_headers = TRUE;
ogg_mux->delta_pad = NULL;
ogg_mux->offset = 0;
ogg_mux->next_ts = 0;
ogg_mux->last_ts = GST_CLOCK_TIME_NONE;
}
static void
gst_ogg_mux_init (GstOggMux * ogg_mux)
{
GstElementClass *klass = GST_ELEMENT_GET_CLASS (ogg_mux);
ogg_mux->srcpad =
gst_pad_new_from_template (gst_element_class_get_pad_template (klass,
"src"), "src");
gst_pad_set_event_function (ogg_mux->srcpad, gst_ogg_mux_handle_src_event);
gst_element_add_pad (GST_ELEMENT (ogg_mux), ogg_mux->srcpad);
renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition Original commit message from CVS: * examples/indexing/indexmpeg.c: (main): * ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get), (cdparanoia_open), (cdparanoia_close), (cdparanoia_event), (cdparanoia_convert), (cdparanoia_query): * ext/cdparanoia/gstcdparanoia.h: * ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_init), (gst_gnomevfssink_open_file), (gst_gnomevfssink_close_file), (gst_gnomevfssink_chain), (gst_gnomevfssink_change_state): * ext/ogg/gstoggmux.c: (gst_ogg_mux_init): * gst/audioscale/gstaudioscale.c: (gst_audioscale_init): * gst/playback/gststreamselector.c: (gst_stream_selector_init): * gst/tcp/gstmultifdsink.c: (gst_multifdsink_init), (gst_multifdsink_render), (gst_multifdsink_start), (gst_multifdsink_stop): * gst/tcp/gstmultifdsink.h: * gst/tcp/gsttcpclientsink.c: (gst_tcpclientsink_init), (gst_tcpclientsink_render), (gst_tcpclientsink_start), (gst_tcpclientsink_stop): * gst/tcp/gsttcpclientsink.h: * gst/tcp/gsttcpclientsrc.c: (gst_tcpclientsrc_init), (gst_tcpclientsrc_getcaps), (gst_tcpclientsrc_create), (gst_tcpclientsrc_start), (gst_tcpclientsrc_stop): * gst/tcp/gsttcpclientsrc.h: * gst/tcp/gsttcpserversink.h: * gst/tcp/gsttcpserversrc.c: (gst_tcpserversrc_init), (gst_tcpserversrc_create), (gst_tcpserversrc_start), (gst_tcpserversrc_stop): * gst/tcp/gsttcpserversrc.h: * sys/v4l/gstv4lmjpegsink.c: (gst_v4lmjpegsink_init): * sys/v4l/gstv4lmjpegsrc.c: (gst_v4lmjpegsrc_init): * sys/ximage/ximagesink.c: (gst_ximagesink_init): renamed GST_FLAGS macros to GST_OBJECT_FLAGS moved bitshift from macro to enum definition
2005-10-12 14:32:29 +00:00
GST_OBJECT_FLAG_SET (GST_ELEMENT (ogg_mux), GST_OGG_FLAG_BOS);
/* seed random number generator for creation of serial numbers */
srand (time (NULL));
ogg_mux->collect = gst_collect_pads_new ();
gst_collect_pads_set_function (ogg_mux->collect,
(GstCollectPadsFunction) GST_DEBUG_FUNCPTR (gst_ogg_mux_collected),
ogg_mux);
ogg_mux->max_delay = DEFAULT_MAX_DELAY;
ogg_mux->max_page_delay = DEFAULT_MAX_PAGE_DELAY;
ogg_mux->max_tolerance = DEFAULT_MAX_TOLERANCE;
gst_ogg_mux_clear (ogg_mux);
}
static void
gst_ogg_mux_finalize (GObject * object)
{
GstOggMux *ogg_mux;
ogg_mux = GST_OGG_MUX (object);
if (ogg_mux->collect) {
gst_object_unref (ogg_mux->collect);
ogg_mux->collect = NULL;
}
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static void
gst_ogg_mux_ogg_pad_destroy_notify (GstCollectData * data)
{
GstOggPadData *oggpad = (GstOggPadData *) data;
GstBuffer *buf;
ogg_stream_clear (&oggpad->map.stream);
2011-02-02 17:30:15 +00:00
gst_caps_replace (&oggpad->map.caps, NULL);
if (oggpad->pagebuffers) {
while ((buf = g_queue_pop_head (oggpad->pagebuffers)) != NULL) {
gst_buffer_unref (buf);
}
g_queue_free (oggpad->pagebuffers);
oggpad->pagebuffers = NULL;
}
}
static GstPadLinkReturn
gst_ogg_mux_sinkconnect (GstPad * pad, GstPad * peer)
{
GstOggMux *ogg_mux;
ogg_mux = GST_OGG_MUX (gst_pad_get_parent (pad));
GST_DEBUG_OBJECT (ogg_mux, "sinkconnect triggered on %s", GST_PAD_NAME (pad));
gst_object_unref (ogg_mux);
return GST_PAD_LINK_OK;
}
static gboolean
gst_ogg_mux_sink_event (GstPad * pad, GstEvent * event)
{
GstOggMux *ogg_mux = GST_OGG_MUX (gst_pad_get_parent (pad));
GstOggPadData *ogg_pad = (GstOggPadData *) gst_pad_get_element_private (pad);
gboolean ret = FALSE;
GST_DEBUG_OBJECT (pad, "Got %s event", GST_EVENT_TYPE_NAME (event));
switch (GST_EVENT_TYPE (event)) {
case GST_EVENT_NEWSEGMENT:{
gboolean update;
gdouble rate;
gdouble applied_rate;
GstFormat format;
gint64 start, stop, position;
gst_event_parse_new_segment_full (event, &update, &rate,
&applied_rate, &format, &start, &stop, &position);
/* We don't support non time NEWSEGMENT events */
if (format != GST_FORMAT_TIME) {
gst_event_unref (event);
event = NULL;
break;
}
gst_segment_set_newsegment_full (&ogg_pad->segment, update, rate,
applied_rate, format, start, stop, position);
break;
}
case GST_EVENT_FLUSH_STOP:{
gst_segment_init (&ogg_pad->segment, GST_FORMAT_TIME);
break;
}
default:
break;
}
/* now GstCollectPads can take care of the rest, e.g. EOS */
if (event != NULL)
ret = ogg_pad->collect_event (pad, event);
gst_object_unref (ogg_mux);
return ret;
}
static gboolean
gst_ogg_mux_is_serialno_present (GstOggMux * ogg_mux, guint32 serialno)
{
GSList *walk;
walk = ogg_mux->collect->data;
while (walk) {
GstOggPadData *pad = (GstOggPadData *) walk->data;
if (pad->map.serialno == serialno)
return TRUE;
walk = walk->next;
}
return FALSE;
}
static guint32
gst_ogg_mux_generate_serialno (GstOggMux * ogg_mux)
{
guint32 serialno;
do {
serialno = g_random_int_range (0, G_MAXINT32);
} while (gst_ogg_mux_is_serialno_present (ogg_mux, serialno));
return serialno;
}
static GstPad *
gst_ogg_mux_request_new_pad (GstElement * element,
GstPadTemplate * templ, const gchar * req_name)
{
GstOggMux *ogg_mux;
GstPad *newpad;
GstElementClass *klass;
g_return_val_if_fail (templ != NULL, NULL);
if (templ->direction != GST_PAD_SINK)
goto wrong_direction;
g_return_val_if_fail (GST_IS_OGG_MUX (element), NULL);
ogg_mux = GST_OGG_MUX (element);
klass = GST_ELEMENT_GET_CLASS (element);
if (templ != gst_element_class_get_pad_template (klass, "sink_%d"))
goto wrong_template;
{
gint serial;
gchar *name;
if (req_name == NULL || strlen (req_name) < 6) {
/* no name given when requesting the pad, use random serial number */
serial = gst_ogg_mux_generate_serialno (ogg_mux);
} else {
/* parse serial number from requested padname */
serial = atoi (&req_name[5]);
}
/* create new pad with the name */
GST_DEBUG_OBJECT (ogg_mux, "Creating new pad for serial %d", serial);
name = g_strdup_printf ("sink_%d", serial);
newpad = gst_pad_new_from_template (templ, name);
g_free (name);
/* construct our own wrapper data structure for the pad to
* keep track of its status */
{
GstOggPadData *oggpad;
oggpad = (GstOggPadData *)
gst_collect_pads_add_pad_full (ogg_mux->collect, newpad,
sizeof (GstOggPadData), gst_ogg_mux_ogg_pad_destroy_notify);
ogg_mux->active_pads++;
oggpad->map.serialno = serial;
ogg_stream_init (&oggpad->map.stream, oggpad->map.serialno);
oggpad->packetno = 0;
oggpad->pageno = 0;
oggpad->eos = FALSE;
/* we assume there will be some control data first for this pad */
oggpad->state = GST_OGG_PAD_STATE_CONTROL;
oggpad->new_page = TRUE;
oggpad->first_delta = FALSE;
oggpad->prev_delta = FALSE;
oggpad->data_pushed = FALSE;
oggpad->pagebuffers = g_queue_new ();
oggpad->map.headers = NULL;
oggpad->map.queued = NULL;
oggpad->next_granule = 0;
oggpad->keyframe_granule = -1;
gst_segment_init (&oggpad->segment, GST_FORMAT_TIME);
oggpad->collect_event = (GstPadEventFunction) GST_PAD_EVENTFUNC (newpad);
gst_pad_set_event_function (newpad,
GST_DEBUG_FUNCPTR (gst_ogg_mux_sink_event));
}
}
/* setup some pad functions */
gst_pad_set_link_function (newpad, gst_ogg_mux_sinkconnect);
/* dd the pad to the element */
gst_element_add_pad (element, newpad);
return newpad;
/* ERRORS */
wrong_direction:
{
g_warning ("ogg_mux: request pad that is not a SINK pad\n");
return NULL;
}
wrong_template:
{
g_warning ("ogg_mux: this is not our template!\n");
return NULL;
}
}
static void
gst_ogg_mux_release_pad (GstElement * element, GstPad * pad)
{
GstOggMux *ogg_mux;
ogg_mux = GST_OGG_MUX (gst_pad_get_parent (pad));
gst_collect_pads_remove_pad (ogg_mux->collect, pad);
gst_element_remove_pad (element, pad);
gst_object_unref (ogg_mux);
}
/* handle events */
static gboolean
gst_ogg_mux_handle_src_event (GstPad * pad, GstEvent * event)
{
GstEventType type;
type = event ? GST_EVENT_TYPE (event) : GST_EVENT_UNKNOWN;
switch (type) {
case GST_EVENT_SEEK:
/* disable seeking for now */
return FALSE;
default:
break;
}
return gst_pad_event_default (pad, event);
}
static GstBuffer *
gst_ogg_mux_buffer_from_page (GstOggMux * mux, ogg_page * page, gboolean delta)
{
GstBuffer *buffer;
/* allocate space for header and body */
buffer = gst_buffer_new_and_alloc (page->header_len + page->body_len);
memcpy (GST_BUFFER_DATA (buffer), page->header, page->header_len);
memcpy (GST_BUFFER_DATA (buffer) + page->header_len,
page->body, page->body_len);
/* Here we set granulepos as our OFFSET_END to give easy direct access to
* this value later. Before we push it, we reset this to OFFSET + SIZE
* (see gst_ogg_mux_push_buffer). */
GST_BUFFER_OFFSET_END (buffer) = ogg_page_granulepos (page);
if (delta)
Port from GstData to GstMiniObject. Original commit message from CVS: Port from GstData to GstMiniObject. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose): * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page), (gst_ogg_mux_queue_pads), (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_collected): * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain): * ext/theora/theoradec.c: (theora_handle_comment_packet), (theora_handle_data_packet): * ext/theora/theoraenc.c: (theora_buffer_from_packet), (theora_set_header_on_caps), (theora_enc_chain): * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event), (vorbis_handle_comment_packet): * ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps): * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps): * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_chain): * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_chain): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_get_buffer): * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered), (mute_stream), (silence_stream): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/volume/gstvolume.c: (volume_transform): * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize), (gst_ximage_buffer_init), (gst_ximage_buffer_class_init), (gst_ximage_buffer_get_type), (gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put), (gst_ximagesink_imagepool_clear), (gst_ximagesink_show_frame), (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc): * sys/ximage/ximagesink.h:
2005-05-16 15:35:52 +00:00
GST_BUFFER_FLAG_SET (buffer, GST_BUFFER_FLAG_DELTA_UNIT);
GST_LOG_OBJECT (mux, GST_GP_FORMAT
" created buffer %p from ogg page",
GST_GP_CAST (ogg_page_granulepos (page)), buffer);
return buffer;
}
static GstFlowReturn
gst_ogg_mux_push_buffer (GstOggMux * mux, GstBuffer * buffer,
GstOggPadData * oggpad)
{
GstCaps *caps;
/* fix up OFFSET and OFFSET_END again */
GST_BUFFER_OFFSET (buffer) = mux->offset;
mux->offset += GST_BUFFER_SIZE (buffer);
GST_BUFFER_OFFSET_END (buffer) = mux->offset;
/* Ensure we have monotonically increasing timestamps in the output. */
if (GST_BUFFER_TIMESTAMP_IS_VALID (buffer)) {
gint64 run_time = GST_BUFFER_TIMESTAMP (buffer);
if (mux->last_ts != GST_CLOCK_TIME_NONE && run_time < mux->last_ts)
GST_BUFFER_TIMESTAMP (buffer) = mux->last_ts;
else
mux->last_ts = run_time;
}
caps = gst_pad_get_negotiated_caps (mux->srcpad);
gst_buffer_set_caps (buffer, caps);
if (caps)
gst_caps_unref (caps);
return gst_pad_push (mux->srcpad, buffer);
}
/* if all queues have at least one page, dequeue the page with the lowest
* timestamp */
static gboolean
gst_ogg_mux_dequeue_page (GstOggMux * mux, GstFlowReturn * flowret)
{
GSList *walk;
GstOggPadData *opad = NULL; /* "oldest" pad */
GstClockTime oldest = GST_CLOCK_TIME_NONE;
GstBuffer *buf = NULL;
gboolean ret = FALSE;
*flowret = GST_FLOW_OK;
walk = mux->collect->data;
while (walk) {
GstOggPadData *pad = (GstOggPadData *) walk->data;
/* We need each queue to either be at EOS, or have one or more pages
* available with a set granulepos (i.e. not -1), otherwise we don't have
* enough data yet to determine which stream needs to go next for correct
* time ordering. */
if (pad->pagebuffers->length == 0) {
if (pad->eos) {
GST_LOG_OBJECT (pad->collect.pad,
"pad is EOS, skipping for dequeue decision");
} else {
GST_LOG_OBJECT (pad->collect.pad,
"no pages in this queue, can't dequeue");
return FALSE;
}
} else {
/* We then need to check for a non-negative granulepos */
int i;
gboolean valid = FALSE;
for (i = 0; i < pad->pagebuffers->length; i++) {
buf = g_queue_peek_nth (pad->pagebuffers, i);
/* Here we check the OFFSET_END, which is actually temporarily the
* granulepos value for this buffer */
if (GST_BUFFER_OFFSET_END (buf) != -1) {
valid = TRUE;
break;
}
}
if (!valid) {
GST_LOG_OBJECT (pad->collect.pad,
"No page timestamps in queue, can't dequeue");
return FALSE;
}
}
walk = g_slist_next (walk);
}
walk = mux->collect->data;
while (walk) {
GstOggPadData *pad = (GstOggPadData *) walk->data;
/* any page with a granulepos of -1 can be pushed immediately.
* TODO: it CAN be, but it seems silly to do so? */
buf = g_queue_peek_head (pad->pagebuffers);
while (buf && GST_BUFFER_OFFSET_END (buf) == -1) {
GST_LOG_OBJECT (pad->collect.pad, "[gp -1] pushing page");
g_queue_pop_head (pad->pagebuffers);
*flowret = gst_ogg_mux_push_buffer (mux, buf, pad);
buf = g_queue_peek_head (pad->pagebuffers);
ret = TRUE;
}
if (buf) {
/* if no oldest buffer yet, take this one */
if (oldest == GST_CLOCK_TIME_NONE) {
GST_LOG_OBJECT (mux, "no oldest yet, taking buffer %p from pad %"
GST_PTR_FORMAT " with gp time %" GST_TIME_FORMAT,
buf, pad->collect.pad, GST_TIME_ARGS (GST_BUFFER_OFFSET (buf)));
oldest = GST_BUFFER_OFFSET (buf);
opad = pad;
} else {
/* if we have an oldest, compare with this one */
if (GST_BUFFER_OFFSET (buf) < oldest) {
GST_LOG_OBJECT (mux, "older buffer %p, taking from pad %"
GST_PTR_FORMAT " with gp time %" GST_TIME_FORMAT,
buf, pad->collect.pad, GST_TIME_ARGS (GST_BUFFER_OFFSET (buf)));
oldest = GST_BUFFER_OFFSET (buf);
opad = pad;
}
}
}
walk = g_slist_next (walk);
}
if (oldest != GST_CLOCK_TIME_NONE) {
g_assert (opad);
buf = g_queue_pop_head (opad->pagebuffers);
GST_LOG_OBJECT (opad->collect.pad,
GST_GP_FORMAT " pushing oldest page buffer %p (granulepos time %"
GST_TIME_FORMAT ")", GST_BUFFER_OFFSET_END (buf), buf,
GST_TIME_ARGS (GST_BUFFER_OFFSET (buf)));
*flowret = gst_ogg_mux_push_buffer (mux, buf, opad);
ret = TRUE;
}
return ret;
}
/* put the given ogg page on a per-pad queue, timestamping it correctly.
* after that, dequeue and push as many pages as possible.
* Caller should make sure:
* pad->timestamp was set with the timestamp of the first packet put
* on the page
* pad->timestamp_end was set with the timestamp + duration of the last packet
* put on the page
* pad->gp_time was set with the time matching the gp of the last
* packet put on the page
*
* will also reset timestamp and timestamp_end, so caller func can restart
* counting.
*/
static GstFlowReturn
gst_ogg_mux_pad_queue_page (GstOggMux * mux, GstOggPadData * pad,
ogg_page * page, gboolean delta)
{
GstFlowReturn ret;
GstBuffer *buffer = gst_ogg_mux_buffer_from_page (mux, page, delta);
/* take the timestamp of the first packet on this page */
GST_BUFFER_TIMESTAMP (buffer) = pad->timestamp;
GST_BUFFER_DURATION (buffer) = pad->timestamp_end - pad->timestamp;
/* take the gp time of the last completed packet on this page */
GST_BUFFER_OFFSET (buffer) = pad->gp_time;
/* the next page will start where the current page's end time leaves off */
pad->timestamp = pad->timestamp_end;
g_queue_push_tail (pad->pagebuffers, buffer);
GST_LOG_OBJECT (pad->collect.pad, GST_GP_FORMAT
" queued buffer page %p (gp time %"
GST_TIME_FORMAT ", timestamp %" GST_TIME_FORMAT
"), %d page buffers queued", GST_GP_CAST (ogg_page_granulepos (page)),
buffer, GST_TIME_ARGS (GST_BUFFER_OFFSET (buffer)),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buffer)),
g_queue_get_length (pad->pagebuffers));
while (gst_ogg_mux_dequeue_page (mux, &ret)) {
if (ret != GST_FLOW_OK)
break;
}
return ret;
}
/*
* Given two pads, compare the buffers queued on it.
* Returns:
* 0 if they have an equal priority
* -1 if the first is better
* 1 if the second is better
* Priority decided by: a) validity, b) older timestamp, c) smaller number
* of muxed pages
*/
static gint
gst_ogg_mux_compare_pads (GstOggMux * ogg_mux, GstOggPadData * first,
GstOggPadData * second)
{
guint64 firsttime, secondtime;
/* if the first pad doesn't contain anything or is even NULL, return
* the second pad as best candidate and vice versa */
if (first == NULL)
return 1;
if (second == NULL)
return -1;
/* no timestamp on first buffer, it must go first */
firsttime = GST_BUFFER_TIMESTAMP (first->buffer);
if (firsttime == GST_CLOCK_TIME_NONE)
return -1;
/* no timestamp on second buffer, it must go first */
secondtime = GST_BUFFER_TIMESTAMP (second->buffer);
if (secondtime == GST_CLOCK_TIME_NONE)
return 1;
/* first buffer has higher timestamp, second one should go first */
if (secondtime < firsttime)
return 1;
/* second buffer has higher timestamp, first one should go first */
else if (secondtime > firsttime)
return -1;
else {
/* buffers with equal timestamps, prefer the pad that has the
* least number of pages muxed */
if (second->pageno < first->pageno)
return 1;
else if (second->pageno > first->pageno)
return -1;
}
/* same priority if all of the above failed */
return 0;
}
static GstBuffer *
gst_ogg_mux_decorate_buffer (GstOggMux * ogg_mux, GstOggPadData * pad,
GstBuffer * buf)
{
GstClockTime time;
gint64 duration, granule, limit;
GstClockTime next_time;
GstClockTimeDiff diff;
ogg_packet packet;
/* ensure messing with metadata is ok */
buf = gst_buffer_make_metadata_writable (buf);
/* convert time to running time, so we need no longer bother about that */
time = GST_BUFFER_TIMESTAMP (buf);
if (G_LIKELY (GST_CLOCK_TIME_IS_VALID (time))) {
time = gst_segment_to_running_time (&pad->segment, GST_FORMAT_TIME, time);
if (G_UNLIKELY (!GST_CLOCK_TIME_IS_VALID (time))) {
gst_buffer_unref (buf);
return NULL;
} else {
GST_BUFFER_TIMESTAMP (buf) = time;
}
}
/* now come up with granulepos stuff corresponding to time */
if (!pad->have_type ||
pad->map.granulerate_n <= 0 || pad->map.granulerate_d <= 0)
goto no_granule;
packet.packet = GST_BUFFER_DATA (buf);
packet.bytes = GST_BUFFER_SIZE (buf);
duration = gst_ogg_stream_get_packet_duration (&pad->map, &packet);
/* give up if no duration can be determined, relying on upstream */
if (G_UNLIKELY (duration < 0)) {
/* well, if some day we really could handle sparse input ... */
if (pad->map.is_sparse) {
limit = 1;
diff = 2;
goto resync;
}
GST_WARNING_OBJECT (pad->collect.pad,
"failed to determine packet duration");
goto no_granule;
}
GST_LOG_OBJECT (pad->collect.pad, "buffer ts %" GST_TIME_FORMAT
", duration %" GST_TIME_FORMAT ", granule duration %" G_GINT64_FORMAT,
GST_TIME_ARGS (time), GST_TIME_ARGS (GST_BUFFER_DURATION (buf)),
duration);
/* determine granule corresponding to time,
* using the inverse of oggdemux' granule -> time */
/* see if interpolated granule matches good enough */
granule = pad->next_granule;
next_time = gst_ogg_stream_granule_to_time (&pad->map, pad->next_granule);
diff = GST_CLOCK_DIFF (next_time, time);
/* we tolerate deviation up to configured or within granule granularity */
limit = gst_ogg_stream_granule_to_time (&pad->map, 1) / 2;
limit = MAX (limit, ogg_mux->max_tolerance);
GST_LOG_OBJECT (pad->collect.pad, "expected granule %" G_GINT64_FORMAT " == "
"time %" GST_TIME_FORMAT " --> ts diff %" GST_TIME_FORMAT
" < tolerance %" GST_TIME_FORMAT " (?)",
granule, GST_TIME_ARGS (next_time), GST_TIME_ARGS (ABS (diff)),
GST_TIME_ARGS (limit));
resync:
/* if not good enough, determine granule based on time */
if (diff > limit || diff < -limit) {
granule = gst_util_uint64_scale_round (time, pad->map.granulerate_n,
GST_SECOND * pad->map.granulerate_d);
GST_DEBUG_OBJECT (pad->collect.pad,
"resyncing to determined granule %" G_GINT64_FORMAT, granule);
}
if (pad->map.is_ogm || pad->map.is_sparse) {
pad->next_granule = granule;
} else {
granule += duration;
pad->next_granule = granule;
}
/* track previous keyframe */
if (!GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DELTA_UNIT))
pad->keyframe_granule = granule;
/* determine corresponding time and granulepos */
GST_BUFFER_OFFSET (buf) = gst_ogg_stream_granule_to_time (&pad->map, granule);
GST_BUFFER_OFFSET_END (buf) =
gst_ogg_stream_granule_to_granulepos (&pad->map, granule,
pad->keyframe_granule);
return buf;
/* ERRORS */
no_granule:
{
GST_DEBUG_OBJECT (pad->collect.pad, "could not determine granulepos, "
"falling back to upstream provided metadata");
return buf;
}
}
/* make sure at least one buffer is queued on all pads, two if possible
*
* if pad->buffer == NULL, pad->next_buffer != NULL, then
* we do not know if the buffer is the last or not
* if pad->buffer != NULL, pad->next_buffer != NULL, then
* pad->buffer is not the last buffer for the pad
* if pad->buffer != NULL, pad->next_buffer == NULL, then
* pad->buffer if the last buffer for the pad
*
* returns a pointer to an oggpad that holds the best buffer, or
* NULL when no pad was usable. "best" means the buffer marked
* with the lowest timestamp. If best->buffer == NULL then either
* we're at EOS (popped = FALSE), or a buffer got dropped, so retry. */
static GstOggPadData *
gst_ogg_mux_queue_pads (GstOggMux * ogg_mux, gboolean * popped)
{
GstOggPadData *bestpad = NULL;
GSList *walk;
*popped = FALSE;
/* try to make sure we have a buffer from each usable pad first */
walk = ogg_mux->collect->data;
while (walk) {
GstOggPadData *pad;
GstCollectData *data;
data = (GstCollectData *) walk->data;
pad = (GstOggPadData *) data;
walk = g_slist_next (walk);
GST_LOG_OBJECT (data->pad, "looking at pad for buffer");
/* try to get a new buffer for this pad if needed and possible */
if (pad->buffer == NULL) {
GstBuffer *buf;
buf = gst_collect_pads_pop (ogg_mux->collect, data);
GST_LOG_OBJECT (data->pad, "popped buffer %" GST_PTR_FORMAT, buf);
/* On EOS we get a NULL buffer */
if (buf != NULL) {
*popped = TRUE;
if (ogg_mux->delta_pad == NULL &&
GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DELTA_UNIT))
ogg_mux->delta_pad = pad;
/* if we need headers */
if (pad->state == GST_OGG_PAD_STATE_CONTROL) {
/* and we have one */
ogg_packet packet;
gboolean is_header;
packet.packet = GST_BUFFER_DATA (buf);
packet.bytes = GST_BUFFER_SIZE (buf);
/* if we're not yet in data mode, ensure we're setup on the first packet */
if (!pad->have_type) {
/* Use headers in caps, if any; this will allow us to be resilient
* to starting streams on the fly, and some streams (like VP8
* at least) do not send headers packets, as other muxers don't
* expect/need them. */
pad->have_type =
gst_ogg_stream_setup_map_from_caps_headers (&pad->map,
GST_BUFFER_CAPS (buf));
if (!pad->have_type) {
/* fallback on the packet */
pad->have_type = gst_ogg_stream_setup_map (&pad->map, &packet);
}
if (!pad->have_type) {
GST_ERROR_OBJECT (pad, "mapper didn't recognise input stream "
"(pad caps: %" GST_PTR_FORMAT ")", GST_PAD_CAPS (pad));
} else {
GST_DEBUG_OBJECT (pad, "caps detected: %" GST_PTR_FORMAT,
pad->map.caps);
}
}
if (pad->have_type)
is_header = gst_ogg_stream_packet_is_header (&pad->map, &packet);
else /* fallback (FIXME 0.11: remove IN_CAPS hack) */
is_header = GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
if (is_header) {
GST_DEBUG_OBJECT (ogg_mux,
"got header buffer in control state, ignoring");
/* just ignore */
pad->map.n_header_packets_seen++;
gst_buffer_unref (buf);
buf = NULL;
} else {
GST_DEBUG_OBJECT (ogg_mux,
"got data buffer in control state, switching to data mode");
/* this is a data buffer so switch to data state */
pad->state = GST_OGG_PAD_STATE_DATA;
/* check if this type of stream allows generating granulepos
* metadata here, if not, upstream will have to provide */
if (gst_ogg_stream_granule_to_granulepos (&pad->map, 1, 1) < 0) {
GST_WARNING_OBJECT (data->pad, "can not generate metadata; "
"relying on upstream");
/* disable metadata code path, otherwise not used anyway */
pad->map.granulerate_n = 0;
}
}
}
/* so now we should have a real data packet;
* see that it is properly decorated */
if (G_LIKELY (buf)) {
buf = gst_ogg_mux_decorate_buffer (ogg_mux, pad, buf);
if (G_UNLIKELY (!buf))
GST_DEBUG_OBJECT (data->pad, "buffer clipped");
}
}
pad->buffer = buf;
}
/* we should have a buffer now, see if it is the best pad to
* pull on */
if (pad->buffer) {
if (gst_ogg_mux_compare_pads (ogg_mux, bestpad, pad) > 0) {
GST_LOG_OBJECT (data->pad,
"new best pad, with buffer %" GST_PTR_FORMAT, pad->buffer);
bestpad = pad;
}
}
}
return bestpad;
}
static GList *
gst_ogg_mux_get_headers (GstOggPadData * pad)
{
GList *res = NULL;
GstStructure *structure;
GstCaps *caps;
GstPad *thepad;
thepad = pad->collect.pad;
GST_LOG_OBJECT (thepad, "getting headers");
caps = gst_pad_get_negotiated_caps (thepad);
if (caps != NULL) {
const GValue *streamheader;
structure = gst_caps_get_structure (caps, 0);
streamheader = gst_structure_get_value (structure, "streamheader");
if (streamheader != NULL) {
GST_LOG_OBJECT (thepad, "got header");
if (G_VALUE_TYPE (streamheader) == GST_TYPE_ARRAY) {
GArray *bufarr = g_value_peek_pointer (streamheader);
gint i;
GST_LOG_OBJECT (thepad, "got fixed list");
for (i = 0; i < bufarr->len; i++) {
GValue *bufval = &g_array_index (bufarr, GValue, i);
GST_LOG_OBJECT (thepad, "item %d", i);
if (G_VALUE_TYPE (bufval) == GST_TYPE_BUFFER) {
GstBuffer *buf = g_value_peek_pointer (bufval);
GST_LOG_OBJECT (thepad, "adding item %d to header list", i);
gst_buffer_ref (buf);
res = g_list_append (res, buf);
}
}
} else {
GST_LOG_OBJECT (thepad, "streamheader is not fixed list");
}
/* Start a new page for every CMML buffer */
if (gst_structure_has_name (structure, "text/x-cmml"))
pad->always_flush_page = TRUE;
} else if (gst_structure_has_name (structure, "video/x-dirac")) {
res = g_list_append (res, pad->buffer);
pad->buffer = NULL;
pad->always_flush_page = TRUE;
} else {
GST_LOG_OBJECT (thepad, "caps don't have streamheader");
}
gst_caps_unref (caps);
} else {
GST_LOG_OBJECT (thepad, "got empty caps as negotiated format");
}
return res;
}
static GstCaps *
gst_ogg_mux_set_header_on_caps (GstCaps * caps, GList * buffers)
{
GstStructure *structure;
GValue array = { 0 };
GList *walk = buffers;
caps = gst_caps_make_writable (caps);
structure = gst_caps_get_structure (caps, 0);
/* put buffers in a fixed list */
g_value_init (&array, GST_TYPE_ARRAY);
while (walk) {
GstBuffer *buf = GST_BUFFER (walk->data);
GstBuffer *copy;
GValue value = { 0 };
walk = walk->next;
/* mark buffer */
GST_LOG ("Setting IN_CAPS on buffer of length %d", GST_BUFFER_SIZE (buf));
Port from GstData to GstMiniObject. Original commit message from CVS: Port from GstData to GstMiniObject. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose): * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page), (gst_ogg_mux_queue_pads), (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_collected): * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain): * ext/theora/theoradec.c: (theora_handle_comment_packet), (theora_handle_data_packet): * ext/theora/theoraenc.c: (theora_buffer_from_packet), (theora_set_header_on_caps), (theora_enc_chain): * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event), (vorbis_handle_comment_packet): * ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps): * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps): * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_chain): * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_chain): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_get_buffer): * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered), (mute_stream), (silence_stream): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/volume/gstvolume.c: (volume_transform): * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize), (gst_ximage_buffer_init), (gst_ximage_buffer_class_init), (gst_ximage_buffer_get_type), (gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put), (gst_ximagesink_imagepool_clear), (gst_ximagesink_show_frame), (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc): * sys/ximage/ximagesink.h:
2005-05-16 15:35:52 +00:00
GST_BUFFER_FLAG_SET (buf, GST_BUFFER_FLAG_IN_CAPS);
g_value_init (&value, GST_TYPE_BUFFER);
copy = gst_buffer_copy (buf);
gst_value_set_buffer (&value, copy);
gst_buffer_unref (copy);
gst_value_array_append_value (&array, &value);
g_value_unset (&value);
}
gst_structure_set_value (structure, "streamheader", &array);
g_value_unset (&array);
return caps;
}
/*
* For each pad we need to write out one (small) header in one
* page that allows decoders to identify the type of the stream.
* After that we need to write out all extra info for the decoders.
* In the case of a codec that also needs data as configuration, we can
* find that info in the streamcaps.
* After writing the headers we must start a new page for the data.
*/
static GstFlowReturn
gst_ogg_mux_send_headers (GstOggMux * mux)
{
GSList *walk;
GList *hbufs, *hwalk;
GstCaps *caps;
GstFlowReturn ret;
hbufs = NULL;
ret = GST_FLOW_OK;
GST_LOG_OBJECT (mux, "collecting headers");
walk = mux->collect->data;
while (walk) {
GstOggPadData *pad;
GstPad *thepad;
pad = (GstOggPadData *) walk->data;
thepad = pad->collect.pad;
walk = g_slist_next (walk);
GST_LOG_OBJECT (mux, "looking at pad %s:%s", GST_DEBUG_PAD_NAME (thepad));
/* if the pad has no buffer, we don't care */
if (pad->buffer == NULL)
continue;
/* now figure out the headers */
pad->map.headers = gst_ogg_mux_get_headers (pad);
}
GST_LOG_OBJECT (mux, "creating BOS pages");
walk = mux->collect->data;
while (walk) {
GstOggPadData *pad;
GstBuffer *buf;
ogg_packet packet;
ogg_page page;
GstPad *thepad;
GstCaps *caps;
GstStructure *structure;
GstBuffer *hbuf;
pad = (GstOggPadData *) walk->data;
thepad = pad->collect.pad;
caps = gst_pad_get_negotiated_caps (thepad);
structure = gst_caps_get_structure (caps, 0);
walk = walk->next;
pad->packetno = 0;
GST_LOG_OBJECT (thepad, "looping over headers");
if (pad->map.headers) {
buf = GST_BUFFER (pad->map.headers->data);
pad->map.headers = g_list_remove (pad->map.headers, buf);
} else if (pad->buffer) {
buf = pad->buffer;
gst_buffer_ref (buf);
} else {
/* fixme -- should be caught in the previous list traversal. */
GST_OBJECT_LOCK (thepad);
g_critical ("No headers or buffers on pad %s:%s",
GST_DEBUG_PAD_NAME (thepad));
GST_OBJECT_UNLOCK (thepad);
continue;
}
/* create a packet from the buffer */
packet.packet = GST_BUFFER_DATA (buf);
packet.bytes = GST_BUFFER_SIZE (buf);
packet.granulepos = GST_BUFFER_OFFSET_END (buf);
if (packet.granulepos == -1)
packet.granulepos = 0;
/* mark BOS and packet number */
packet.b_o_s = (pad->packetno == 0);
packet.packetno = pad->packetno++;
/* mark EOS */
packet.e_o_s = 0;
/* swap the packet in */
ogg_stream_packetin (&pad->map.stream, &packet);
gst_buffer_unref (buf);
GST_LOG_OBJECT (thepad, "flushing out BOS page");
if (!ogg_stream_flush (&pad->map.stream, &page))
g_critical ("Could not flush BOS page");
hbuf = gst_ogg_mux_buffer_from_page (mux, &page, FALSE);
GST_LOG_OBJECT (mux, "swapped out page with mime type %s",
gst_structure_get_name (structure));
2010-05-05 11:59:57 +00:00
/* quick hack: put Theora, VP8 and Dirac video pages at the front.
* Ideally, we would have a settable enum for which Ogg
* profile we work with, and order based on that.
* (FIXME: if there is more than one video stream, shouldn't we only put
* one's BOS into the first page, followed by an audio stream's BOS, and
* only then followed by the remaining video and audio streams?) */
if (gst_structure_has_name (structure, "video/x-theora")) {
GST_DEBUG_OBJECT (thepad, "putting %s page at the front", "Theora");
hbufs = g_list_prepend (hbufs, hbuf);
} else if (gst_structure_has_name (structure, "video/x-dirac")) {
GST_DEBUG_OBJECT (thepad, "putting %s page at the front", "Dirac");
hbufs = g_list_prepend (hbufs, hbuf);
pad->always_flush_page = TRUE;
2010-05-05 11:59:57 +00:00
} else if (gst_structure_has_name (structure, "video/x-vp8")) {
GST_DEBUG_OBJECT (thepad, "putting %s page at the front", "VP8");
hbufs = g_list_prepend (hbufs, hbuf);
} else {
hbufs = g_list_append (hbufs, hbuf);
}
gst_caps_unref (caps);
}
GST_LOG_OBJECT (mux, "creating next headers");
walk = mux->collect->data;
while (walk) {
GstOggPadData *pad;
GstPad *thepad;
pad = (GstOggPadData *) walk->data;
thepad = pad->collect.pad;
walk = walk->next;
GST_LOG_OBJECT (mux, "looping over headers for pad %s:%s",
GST_DEBUG_PAD_NAME (thepad));
hwalk = pad->map.headers;
while (hwalk) {
GstBuffer *buf = GST_BUFFER (hwalk->data);
ogg_packet packet;
ogg_page page;
hwalk = hwalk->next;
/* create a packet from the buffer */
packet.packet = GST_BUFFER_DATA (buf);
packet.bytes = GST_BUFFER_SIZE (buf);
packet.granulepos = GST_BUFFER_OFFSET_END (buf);
if (packet.granulepos == -1)
packet.granulepos = 0;
/* mark BOS and packet number */
packet.b_o_s = (pad->packetno == 0);
packet.packetno = pad->packetno++;
/* mark EOS */
packet.e_o_s = 0;
/* swap the packet in */
ogg_stream_packetin (&pad->map.stream, &packet);
gst_buffer_unref (buf);
/* if last header, flush page */
if (hwalk == NULL) {
GST_LOG_OBJECT (mux,
Printf format fixes. Original commit message from CVS: * ext/alsa/gstalsadeviceprobe.c: (gst_alsa_device_property_probe_get_values): * ext/alsa/gstalsasink.c: (set_hwparams): * ext/ogg/gstoggdemux.c: (gst_ogg_demux_chain_elem_pad), (gst_ogg_chain_new_stream), (gst_ogg_demux_read_chain): * ext/ogg/gstoggmux.c: (gst_ogg_mux_send_headers), (gst_ogg_mux_process_best_pad): * ext/ogg/gstoggparse.c: (gst_ogg_parse_new_stream), (gst_ogg_parse_chain): * ext/ogg/gstogmparse.c: (gst_ogm_parse_stream_header): * ext/vorbis/vorbisdec.c: (vorbis_handle_data_packet): * ext/vorbis/vorbisenc.c: (gst_vorbis_enc_setup), (gst_vorbis_enc_buffer_check_discontinuous): * ext/vorbis/vorbisparse.c: (vorbis_parse_src_query): * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render): * gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_handle_track_seek): * gst-libs/gst/rtp/gstbasertpdepayload.c: (gst_base_rtp_depayload_push_full): * gst-libs/gst/rtp/gstbasertppayload.c: (gst_basertppayload_push): * gst/audioresample/resample.c: (resample_input_pushthrough): * gst/playback/gstplaybasebin.c: (queue_out_of_data): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_handle_clients): * gst/typefind/gsttypefindfunctions.c: (mp3_type_find_at_offset), (wavpack_type_find): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_create): * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy), (gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new): * tests/check/elements/volume.c: (GST_START_TEST): Printf format fixes.
2006-10-05 15:55:21 +00:00
"flushing page as packet %" G_GUINT64_FORMAT " is first or "
"last packet", (guint64) packet.packetno);
while (ogg_stream_flush (&pad->map.stream, &page)) {
GstBuffer *hbuf = gst_ogg_mux_buffer_from_page (mux, &page, FALSE);
GST_LOG_OBJECT (mux, "swapped out page");
hbufs = g_list_append (hbufs, hbuf);
}
} else {
GST_LOG_OBJECT (mux, "try to swap out page");
/* just try to swap out a page then */
while (ogg_stream_pageout (&pad->map.stream, &page) > 0) {
GstBuffer *hbuf = gst_ogg_mux_buffer_from_page (mux, &page, FALSE);
GST_LOG_OBJECT (mux, "swapped out page");
hbufs = g_list_append (hbufs, hbuf);
}
}
}
g_list_free (pad->map.headers);
pad->map.headers = NULL;
}
/* hbufs holds all buffers for the headers now */
/* create caps with the buffers */
caps = gst_pad_get_caps (mux->srcpad);
if (caps) {
caps = gst_ogg_mux_set_header_on_caps (caps, hbufs);
gst_pad_set_caps (mux->srcpad, caps);
gst_caps_unref (caps);
}
/* and send the buffers */
while (hbufs != NULL) {
GstBuffer *buf = GST_BUFFER (hbufs->data);
hbufs = g_list_delete_link (hbufs, hbufs);
if ((ret = gst_ogg_mux_push_buffer (mux, buf, NULL)) != GST_FLOW_OK)
break;
}
/* free any remaining nodes/buffers in case we couldn't push them */
g_list_foreach (hbufs, (GFunc) gst_mini_object_unref, NULL);
g_list_free (hbufs);
return ret;
}
/* this function is called to process data on the best pending pad.
*
* basic idea:
*
* 1) store the selected pad and keep on pulling until we fill a
* complete ogg page or the ogg page is filled above the max-delay
* threshold. This is needed because the ogg spec says that
* you should fill a complete page with data from the same logical
* stream. When the page is filled, go back to 1).
* 2) before filling a page, read ahead one more buffer to see if this
* packet is the last of the stream. We need to do this because the ogg
* spec mandates that the last packet should have the EOS flag set before
* sending it to ogg. if pad->buffer is NULL we need to wait to find out
* whether there are any more buffers.
* 3) pages get queued on a per-pad queue. Every time a page is queued, a
* dequeue is called, which will dequeue the oldest page on any pad, provided
* that ALL pads have at least one marked page in the queue (or remaining
* pads are at EOS)
*/
static GstFlowReturn
gst_ogg_mux_process_best_pad (GstOggMux * ogg_mux, GstOggPadData * best)
{
GstFlowReturn ret = GST_FLOW_OK;
gboolean delta_unit;
gint64 granulepos = 0;
GstClockTime timestamp, gp_time;
GstBuffer *next_buf;
GST_LOG_OBJECT (ogg_mux, "best pad %" GST_PTR_FORMAT
", currently pulling from %" GST_PTR_FORMAT, best->collect.pad,
ogg_mux->pulling ? ogg_mux->pulling->collect.pad : NULL);
if (ogg_mux->pulling) {
next_buf = gst_collect_pads_peek (ogg_mux->collect,
&ogg_mux->pulling->collect);
if (next_buf) {
ogg_mux->pulling->eos = FALSE;
gst_buffer_unref (next_buf);
} else {
GST_DEBUG_OBJECT (ogg_mux->pulling->collect.pad, "setting eos to true");
ogg_mux->pulling->eos = TRUE;
}
}
/* We could end up pushing from the best pad instead, so check that
* as well */
if (best && best != ogg_mux->pulling) {
next_buf = gst_collect_pads_peek (ogg_mux->collect, &best->collect);
if (next_buf) {
best->eos = FALSE;
gst_buffer_unref (next_buf);
} else {
GST_DEBUG_OBJECT (best->collect.pad, "setting eos to true");
best->eos = TRUE;
}
}
/* if we were already pulling from one pad, but the new "best" buffer is
* from another pad, we need to check if we have reason to flush a page
* for the pad we were pulling from before */
if (ogg_mux->pulling && best &&
ogg_mux->pulling != best && ogg_mux->pulling->buffer) {
GstOggPadData *pad = ogg_mux->pulling;
GstClockTime last_ts = GST_BUFFER_END_TIME (pad->buffer);
/* if the next packet in the current page is going to make the page
* too long, we need to flush */
if (last_ts > ogg_mux->next_ts + ogg_mux->max_delay) {
ogg_page page;
GST_LOG_OBJECT (pad->collect.pad,
GST_GP_FORMAT " stored packet %" G_GINT64_FORMAT
" will make page too long, flushing",
GST_BUFFER_OFFSET_END (pad->buffer),
(gint64) pad->map.stream.packetno);
while (ogg_stream_flush (&pad->map.stream, &page)) {
/* end time of this page is the timestamp of the next buffer */
ogg_mux->pulling->timestamp_end = GST_BUFFER_TIMESTAMP (pad->buffer);
/* Place page into the per-pad queue */
ret = gst_ogg_mux_pad_queue_page (ogg_mux, pad, &page,
pad->first_delta);
/* increment the page number counter */
pad->pageno++;
/* mark other pages as delta */
pad->first_delta = TRUE;
}
pad->new_page = TRUE;
ogg_mux->pulling = NULL;
}
}
/* if we don't know which pad to pull on, use the best one */
if (ogg_mux->pulling == NULL) {
ogg_mux->pulling = best;
GST_LOG_OBJECT (ogg_mux->pulling->collect.pad, "pulling from best pad");
/* remember timestamp and gp time of first buffer for this new pad */
if (ogg_mux->pulling != NULL) {
ogg_mux->next_ts = GST_BUFFER_TIMESTAMP (ogg_mux->pulling->buffer);
GST_LOG_OBJECT (ogg_mux->pulling->collect.pad, "updated times, next ts %"
GST_TIME_FORMAT, GST_TIME_ARGS (ogg_mux->next_ts));
} else {
/* no pad to pull on, send EOS */
examples/seeking/seek.c: Update seek example. Original commit message from CVS: * examples/seeking/seek.c: (setup_dynamic_link), (make_dv_pipeline), (make_vorbis_theora_pipeline), (query_rates), (query_positions_elems), (query_positions_pads), (do_seek): Update seek example. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_event), (gst_ogg_pad_typefind), (gst_ogg_demux_chain_elem_pad), (gst_ogg_demux_queue_data), (gst_ogg_demux_chain_peer), (gst_ogg_pad_submit_packet), (gst_ogg_pad_submit_page), (gst_ogg_demux_handle_event), (gst_ogg_demux_deactivate_current_chain), (gst_ogg_demux_activate_chain), (gst_ogg_demux_perform_seek), (gst_ogg_demux_collect_chain_info), (gst_ogg_demux_collect_info), (gst_ogg_demux_chain), (gst_ogg_demux_send_event), (gst_ogg_demux_loop): * ext/ogg/gstoggmux.c: (gst_ogg_mux_collected): * ext/theora/theoradec.c: (theora_dec_src_event), (theora_dec_src_getcaps), (theora_dec_sink_event), (theora_dec_push), (theora_dec_chain): * ext/vorbis/Makefile.am: * ext/vorbis/vorbisdec.c: (vorbis_dec_src_event), (vorbis_dec_sink_event), (vorbis_dec_push), (vorbis_handle_data_packet): * ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_event), (gst_vorbisenc_chain): * gst/playback/gststreaminfo.c: (cb_probe): * gst/subparse/gstsubparse.c: (gst_subparse_src_event): * gst/videorate/gstvideorate.c: (gst_videorate_event): * gst/videoscale/gstvideoscale.c: (gst_videoscale_handle_src_event): * gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_event): * sys/ximage/ximagesink.c: (gst_ximagesink_show_frame), (gst_ximagesink_navigation_send_event): * sys/xvimage/xvimagesink.c: (gst_xvimagesink_navigation_send_event): Various event updates and cleanups
2005-07-27 18:34:29 +00:00
gst_pad_push_event (ogg_mux->srcpad, gst_event_new_eos ());
return GST_FLOW_WRONG_STATE;
}
}
if (ogg_mux->need_headers) {
ret = gst_ogg_mux_send_headers (ogg_mux);
ogg_mux->need_headers = FALSE;
}
/* we are pulling from a pad, continue to do so until a page
* has been filled and queued */
if (ogg_mux->pulling != NULL) {
ogg_packet packet;
ogg_page page;
GstBuffer *buf, *tmpbuf;
GstOggPadData *pad = ogg_mux->pulling;
gint64 duration;
gboolean force_flush;
GST_LOG_OBJECT (ogg_mux->pulling->collect.pad, "pulling from pad");
/* now see if we have a buffer */
buf = pad->buffer;
if (buf == NULL) {
GST_DEBUG_OBJECT (ogg_mux, "pad was EOS");
ogg_mux->pulling = NULL;
return GST_FLOW_OK;
}
Port from GstData to GstMiniObject. Original commit message from CVS: Port from GstData to GstMiniObject. * ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose): * ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page), (gst_ogg_mux_queue_pads), (gst_ogg_mux_set_header_on_caps), (gst_ogg_mux_collected): * ext/ogg/gstogmparse.c: (gst_ogm_parse_chain): * ext/theora/theoradec.c: (theora_handle_comment_packet), (theora_handle_data_packet): * ext/theora/theoraenc.c: (theora_buffer_from_packet), (theora_set_header_on_caps), (theora_enc_chain): * ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event), (vorbis_handle_comment_packet): * ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps): * ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps): * gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_chain): * gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_chain): * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_get_buffer): * gst/playback/gstdecodebin.c: (gst_decode_bin_class_init): * gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered), (mute_stream), (silence_stream): * gst/playback/gstplaybin.c: (gst_play_bin_class_init): * gst/volume/gstvolume.c: (volume_transform): * sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize), (gst_ximage_buffer_init), (gst_ximage_buffer_class_init), (gst_ximage_buffer_get_type), (gst_ximagesink_check_xshm_calls), (gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy), (gst_ximagesink_ximage_put), (gst_ximagesink_imagepool_clear), (gst_ximagesink_show_frame), (gst_ximagesink_buffer_free), (gst_ximagesink_buffer_alloc): * sys/ximage/ximagesink.h:
2005-05-16 15:35:52 +00:00
delta_unit = GST_BUFFER_FLAG_IS_SET (buf, GST_BUFFER_FLAG_DELTA_UNIT);
duration = GST_BUFFER_DURATION (buf);
/* if the current "next timestamp" on the pad is unset, then this is the
* first packet on the new page. Update our pad's page timestamp */
if (ogg_mux->pulling->timestamp == GST_CLOCK_TIME_NONE) {
ogg_mux->pulling->timestamp = GST_BUFFER_TIMESTAMP (buf);
GST_LOG_OBJECT (ogg_mux->pulling->collect.pad,
"updated pad timestamp to %" GST_TIME_FORMAT,
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (buf)));
}
/* create a packet from the buffer */
packet.packet = GST_BUFFER_DATA (buf);
packet.bytes = GST_BUFFER_SIZE (buf);
packet.granulepos = GST_BUFFER_OFFSET_END (buf);
if (packet.granulepos == -1)
packet.granulepos = 0;
/* mark BOS and packet number */
packet.b_o_s = (pad->packetno == 0);
packet.packetno = pad->packetno++;
GST_LOG_OBJECT (pad->collect.pad, GST_GP_FORMAT
" packet %" G_GINT64_FORMAT " (%ld bytes) created from buffer",
GST_GP_CAST (packet.granulepos), (gint64) packet.packetno,
packet.bytes);
packet.e_o_s = ogg_mux->pulling->eos ? 1 : 0;
tmpbuf = NULL;
/* we flush when we see a new keyframe */
force_flush = (pad->prev_delta && !delta_unit) || pad->always_flush_page;
if (duration != -1) {
pad->duration += duration;
/* if page duration exceeds max, flush page */
if (pad->duration > ogg_mux->max_page_delay) {
force_flush = TRUE;
pad->duration = 0;
}
}
if (GST_BUFFER_IS_DISCONT (buf)) {
if (pad->data_pushed) {
GST_LOG_OBJECT (pad->collect.pad, "got discont");
packet.packetno++;
/* No public API for this; hack things in */
pad->map.stream.pageno++;
force_flush = TRUE;
} else {
GST_LOG_OBJECT (pad->collect.pad, "discont at stream start");
}
}
/* flush the currently built page if necessary */
if (force_flush) {
GST_LOG_OBJECT (pad->collect.pad,
GST_GP_FORMAT " forced flush of page before this packet",
GST_BUFFER_OFFSET_END (pad->buffer));
while (ogg_stream_flush (&pad->map.stream, &page)) {
/* end time of this page is the timestamp of the next buffer */
ogg_mux->pulling->timestamp_end = GST_BUFFER_TIMESTAMP (pad->buffer);
ret = gst_ogg_mux_pad_queue_page (ogg_mux, pad, &page,
pad->first_delta);
/* increment the page number counter */
pad->pageno++;
/* mark other pages as delta */
pad->first_delta = TRUE;
}
pad->new_page = TRUE;
}
/* if this is the first packet of a new page figure out the delta flag */
if (pad->new_page) {
if (delta_unit) {
/* mark the page as delta */
pad->first_delta = TRUE;
} else {
/* got a keyframe */
if (ogg_mux->delta_pad == pad) {
/* if we get it on the pad with deltaunits,
* we mark the page as non delta */
pad->first_delta = FALSE;
} else if (ogg_mux->delta_pad != NULL) {
/* if there are pads with delta frames, we
* must mark this one as delta */
pad->first_delta = TRUE;
} else {
pad->first_delta = FALSE;
}
}
pad->new_page = FALSE;
}
/* save key unit to track delta->key unit transitions */
pad->prev_delta = delta_unit;
/* swap the packet in */
if (packet.e_o_s == 1)
GST_DEBUG_OBJECT (pad->collect.pad, "swapping in EOS packet");
if (packet.b_o_s == 1)
GST_DEBUG_OBJECT (pad->collect.pad, "swapping in BOS packet");
ogg_stream_packetin (&pad->map.stream, &packet);
pad->data_pushed = TRUE;
gp_time = GST_BUFFER_OFFSET (pad->buffer);
granulepos = GST_BUFFER_OFFSET_END (pad->buffer);
timestamp = GST_BUFFER_TIMESTAMP (pad->buffer);
GST_LOG_OBJECT (pad->collect.pad,
GST_GP_FORMAT " packet %" G_GINT64_FORMAT ", gp time %"
GST_TIME_FORMAT ", timestamp %" GST_TIME_FORMAT " packetin'd",
granulepos, (gint64) packet.packetno, GST_TIME_ARGS (gp_time),
GST_TIME_ARGS (timestamp));
/* don't need the old buffer anymore */
gst_buffer_unref (pad->buffer);
/* store new readahead buffer */
pad->buffer = tmpbuf;
/* let ogg write out the pages now. The packet we got could end
* up in more than one page so we need to write them all */
if (ogg_stream_pageout (&pad->map.stream, &page) > 0) {
/* we have a new page, so we need to timestamp it correctly.
* if this fresh packet ends on this page, then the page's granulepos
* comes from that packet, and we should set this buffer's timestamp */
GST_LOG_OBJECT (pad->collect.pad,
GST_GP_FORMAT " packet %" G_GINT64_FORMAT ", time %"
GST_TIME_FORMAT ") caused new page",
granulepos, (gint64) packet.packetno, GST_TIME_ARGS (timestamp));
GST_LOG_OBJECT (pad->collect.pad,
GST_GP_FORMAT " new page %ld",
GST_GP_CAST (ogg_page_granulepos (&page)), pad->map.stream.pageno);
if (ogg_page_granulepos (&page) == granulepos) {
/* the packet we streamed in finishes on the current page,
* because the page's granulepos is the granulepos of the last
* packet completed on that page,
* so update the timestamp that we will give to the page */
GST_LOG_OBJECT (pad->collect.pad,
GST_GP_FORMAT
" packet finishes on current page, updating gp time to %"
GST_TIME_FORMAT, granulepos, GST_TIME_ARGS (gp_time));
pad->gp_time = gp_time;
} else {
GST_LOG_OBJECT (pad->collect.pad,
GST_GP_FORMAT
" packet spans beyond current page, keeping old gp time %"
GST_TIME_FORMAT, granulepos, GST_TIME_ARGS (pad->gp_time));
}
/* push the page */
/* end time of this page is the timestamp of the next buffer */
pad->timestamp_end = timestamp;
ret = gst_ogg_mux_pad_queue_page (ogg_mux, pad, &page, pad->first_delta);
pad->pageno++;
/* mark next pages as delta */
pad->first_delta = TRUE;
/* use an inner loop here to flush the remaining pages and
* mark them as delta frames as well */
while (ogg_stream_pageout (&pad->map.stream, &page) > 0) {
if (ogg_page_granulepos (&page) == granulepos) {
/* the page has taken up the new packet completely, which means
* the packet ends the page and we can update the gp time
* before pushing out */
pad->gp_time = gp_time;
}
/* we have a complete page now, we can push the page
* and make sure to pull on a new pad the next time around */
ret = gst_ogg_mux_pad_queue_page (ogg_mux, pad, &page,
pad->first_delta);
/* increment the page number counter */
pad->pageno++;
}
/* need a new page as well */
pad->new_page = TRUE;
pad->duration = 0;
/* we're done pulling on this pad, make sure to choose a new
* pad for pulling in the next iteration */
ogg_mux->pulling = NULL;
}
/* Update the gp time, if necessary, since any future page will have at
* least this gp time.
*/
if (pad->gp_time < gp_time) {
pad->gp_time = gp_time;
GST_LOG_OBJECT (pad->collect.pad,
"Updated running gp time of pad %" GST_PTR_FORMAT
" to %" GST_TIME_FORMAT, pad->collect.pad, GST_TIME_ARGS (gp_time));
}
}
return ret;
}
Add some documentation comments, and some new headers to be scanned. Original commit message from CVS: * docs/plugins/Makefile.am: * 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/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-gio.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/cdparanoia/gstcdparanoiasrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggdemux.h: * ext/ogg/gstoggmux.c: * ext/ogg/gstoggmux.h: * gst/audioconvert/audioconvert.c: * gst/audioconvert/audioconvert.h: * gst/audioconvert/gstaudioconvert.h: * gst/gdp/gstgdpdepay.h: * gst/gdp/gstgdppay.h: * gst/playback/gstdecodebin.c: * gst/playback/gstdecodebin2.c: * gst/playback/gstplaybin.c: * gst/playback/gstplaybin2.c: * gst/playback/gsturidecodebin.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gstmultifdsink.h: * gst/tcp/gsttcp.h: Add some documentation comments, and some new headers to be scanned. Rename some internal enum declarations (audioconvert's DitherType and NoiseShapingType, GstUnitType from the TCP elements) to match the documented GObject type names so that the docs pick them up. Name the playbin2 docs markups properly so they get picked up. They'll need renaming back when/if playbin2 becomes playbin. 100% symbol coverage for the plugin docs, booya.
2008-05-22 22:09:16 +00:00
/* all_pads_eos:
*
* Checks if all pads are EOS'd by peeking.
*
* Returns TRUE if all pads are EOS.
*/
static gboolean
all_pads_eos (GstCollectPads * pads)
{
GSList *walk;
walk = pads->data;
while (walk) {
GstOggPadData *oggpad = (GstOggPadData *) walk->data;
GST_DEBUG_OBJECT (oggpad->collect.pad,
"oggpad %p eos %d", oggpad, oggpad->eos);
if (oggpad->eos == FALSE)
return FALSE;
walk = g_slist_next (walk);
}
return TRUE;
}
/* This function is called when there is data on all pads.
*
* It finds a pad to pull on, this is done by looking at the buffers
* to decide which one to use, and using the 'oldest' one first. It then calls
* gst_ogg_mux_process_best_pad() to process as much data as possible.
*
* If all the pads have received EOS, it flushes out all data by continually
* getting the best pad and calling gst_ogg_mux_process_best_pad() until they
* are all empty, and then sends EOS.
*/
static GstFlowReturn
gst_ogg_mux_collected (GstCollectPads * pads, GstOggMux * ogg_mux)
{
GstOggPadData *best;
GstFlowReturn ret;
gboolean popped;
GST_LOG_OBJECT (ogg_mux, "collected");
/* queue buffers on all pads; find a buffer with the lowest timestamp */
best = gst_ogg_mux_queue_pads (ogg_mux, &popped);
if (popped)
return GST_FLOW_OK;
if (best == NULL || best->buffer == NULL) {
/* This is not supposed to happen */
return GST_FLOW_ERROR;
}
ret = gst_ogg_mux_process_best_pad (ogg_mux, best);
if (best->eos && all_pads_eos (pads)) {
gst_pad_push_event (ogg_mux->srcpad, gst_event_new_eos ());
return GST_FLOW_UNEXPECTED;
}
return ret;
}
static void
gst_ogg_mux_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstOggMux *ogg_mux;
ogg_mux = GST_OGG_MUX (object);
switch (prop_id) {
case ARG_MAX_DELAY:
g_value_set_uint64 (value, ogg_mux->max_delay);
break;
case ARG_MAX_PAGE_DELAY:
g_value_set_uint64 (value, ogg_mux->max_page_delay);
break;
case ARG_MAX_TOLERANCE:
g_value_set_uint64 (value, ogg_mux->max_tolerance);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_ogg_mux_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstOggMux *ogg_mux;
ogg_mux = GST_OGG_MUX (object);
switch (prop_id) {
case ARG_MAX_DELAY:
ogg_mux->max_delay = g_value_get_uint64 (value);
break;
case ARG_MAX_PAGE_DELAY:
ogg_mux->max_page_delay = g_value_get_uint64 (value);
break;
case ARG_MAX_TOLERANCE:
ogg_mux->max_tolerance = g_value_get_uint64 (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
/* reset all variables in the ogg pads. */
static void
gst_ogg_mux_init_collectpads (GstCollectPads * collect)
{
GSList *walk;
walk = collect->data;
while (walk) {
GstOggPadData *oggpad = (GstOggPadData *) walk->data;
ogg_stream_init (&oggpad->map.stream, oggpad->map.serialno);
oggpad->packetno = 0;
oggpad->pageno = 0;
oggpad->eos = FALSE;
/* we assume there will be some control data first for this pad */
oggpad->state = GST_OGG_PAD_STATE_CONTROL;
oggpad->new_page = TRUE;
oggpad->first_delta = FALSE;
oggpad->prev_delta = FALSE;
oggpad->data_pushed = FALSE;
oggpad->pagebuffers = g_queue_new ();
gst_segment_init (&oggpad->segment, GST_FORMAT_TIME);
walk = g_slist_next (walk);
}
}
/* Clear all buffers from the collectpads object */
static void
gst_ogg_mux_clear_collectpads (GstCollectPads * collect)
{
GSList *walk;
for (walk = collect->data; walk; walk = g_slist_next (walk)) {
GstOggPadData *oggpad = (GstOggPadData *) walk->data;
GstBuffer *buf;
ogg_stream_clear (&oggpad->map.stream);
while ((buf = g_queue_pop_head (oggpad->pagebuffers)) != NULL) {
gst_buffer_unref (buf);
}
g_queue_free (oggpad->pagebuffers);
oggpad->pagebuffers = NULL;
if (oggpad->buffer) {
gst_buffer_unref (oggpad->buffer);
oggpad->buffer = NULL;
}
gst_segment_init (&oggpad->segment, GST_FORMAT_TIME);
}
}
static GstStateChangeReturn
gst_ogg_mux_change_state (GstElement * element, GstStateChange transition)
{
GstOggMux *ogg_mux;
GstStateChangeReturn ret;
ogg_mux = GST_OGG_MUX (element);
switch (transition) {
case GST_STATE_CHANGE_NULL_TO_READY:
break;
case GST_STATE_CHANGE_READY_TO_PAUSED:
gst_ogg_mux_clear (ogg_mux);
gst_ogg_mux_init_collectpads (ogg_mux->collect);
gst_collect_pads_start (ogg_mux->collect);
break;
case GST_STATE_CHANGE_PAUSED_TO_PLAYING:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_collect_pads_stop (ogg_mux->collect);
break;
default:
break;
}
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
switch (transition) {
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
break;
case GST_STATE_CHANGE_PAUSED_TO_READY:
gst_ogg_mux_clear_collectpads (ogg_mux->collect);
break;
case GST_STATE_CHANGE_READY_TO_NULL:
break;
default:
break;
}
return ret;
}
gboolean
gst_ogg_mux_plugin_init (GstPlugin * plugin)
{
GST_DEBUG_CATEGORY_INIT (gst_ogg_mux_debug, "oggmux", 0, "ogg muxer");
return gst_element_register (plugin, "oggmux", GST_RANK_PRIMARY,
GST_TYPE_OGG_MUX);
}