Commit f8370edc authored by Jan Schmidt's avatar Jan Schmidt

Release 0.10.23

parent 4d8f38fd
=== release 0.10.23 ===
2009-05-10 Jan Schmidt <jan.schmidt@sun.com>
* configure.ac:
releasing 0.10.23, "Emergency de-stress call"
2009-05-08 20:32:20 +0100 Jan Schmidt <thaytan@noraisin.net>
* configure.ac:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
* win32/common/_stdint.h:
* win32/common/config.h:
0.10.22.6 pre-release
2009-05-08 13:09:32 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/playback/gstplaysink.c:
playbin2: fix resume after pause
Don't ignore the state change of the children, they might be doing an ASYNC
state change.
2009-05-08 11:05:41 +0100 Jan Schmidt <thaytan@noraisin.net>
* ChangeLog:
* configure.ac:
* po/af.po:
* po/az.po:
* po/bg.po:
* po/ca.po:
* po/cs.po:
* po/da.po:
* po/de.po:
* po/en_GB.po:
* po/es.po:
* po/fi.po:
* po/fr.po:
* po/hu.po:
* po/id.po:
* po/it.po:
* po/ja.po:
* po/lt.po:
* po/nb.po:
* po/nl.po:
* po/or.po:
* po/pl.po:
* po/pt_BR.po:
* po/ru.po:
* po/sk.po:
* po/sq.po:
* po/sr.po:
* po/sv.po:
* po/uk.po:
* po/vi.po:
* po/zh_CN.po:
0.10.22.5 pre-release
2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk> 2009-05-07 22:01:01 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
* gst/tcp/gstmultifdsink.c: * gst/tcp/gstmultifdsink.c:
This is GStreamer Base Plug-ins 0.10.22, "Hidey Hidey Hidey Ho" This is GStreamer Base Plug-ins 0.10.23, "Emergency de-stress signal"
Changes since 0.10.22:
* New navigation API to support DVD playback
* playbin2 improvements
* RTSP extensions to allow extra headers and options
* Replace audioresampler with speexresample based code
* Support interlacing flags in the gstvideo library
* Support new RIFF formats
* Improve typefinding
* Support more frame formats in videoscale
* Many other bug-fixes and improvements
Bugs fixed since 0.10.22:
* 577637 : [playbin2] expose temp-location property
* 580120 : [playbin2] unit test fails
* 478512 : [alsamixer] volume control slider not working
* 574962 : rhythmbox crash in flac_type_find
* 564139 : Documentation of TCP plugins
* 577436 : xvimagesink should use xcontext- > depth and not count bits...
* 350311 : [playbin2] support for subpicture subtitles
* 378094 : Enable pango elements to handle UYVY
* 543591 : Gnonlin can not play theora streams
* 553295 : [riff] fuzzed AVI file causes segfault
* 565105 : Gstreamer does not change from READY back to PAUSED in sa...
* 565777 : [riff] unrecognised video fourcc 0x10000002 for mpeg2 in avi
* 566661 : [typefind] Fall back to file extension using uri query
* 567255 : [riff] doesn't detect codec_id 0x706d as AAC (amongst other)
* 567636 : [pbutils] Missing plugins code shouldn't ask for the same...
* 567740 : bogus warning in decodebin2?
* 568482 : linking problems in gst-plugins-base
* 569655 : [ffmpegcolorspace] Add UYVY422 to GRAY8 conversion function
* 570142 : Documentation is broken for uridecodebin
* 570356 : aac typefinder failure
* 570768 : [ximagesink] wrong mouse pointer position if output windo...
* 570832 : Add flags to enhance mixer interfaces
* 571009 : [tagdemux] WMA file with id3v2 tag causes assertion to fail
* 571147 : [ffmpegcolorspace/videotestsrc] Add support for packed/pl...
* 572577 : [playbin2] deadlock on shutdown
* 572872 : [ffmpegcolorspace] Add YVYU colorspace
* 572993 : [subparse] broken libregex dependency on Windows
* 573165 : Generate additional export files for gstreamer app plugin
* 573528 : Wrong format modifier in gstgiobasesink.c
* 573529 : In gstrtspconnection.c some functions are called with wro...
* 574293 : [decodebin2] deadlock on shutdown
* 574319 : Missing HAVE_PROCESS_H in win32/common/config.h
* 574447 : gstadder.c: line 904: error C2036: 'gpointer' : unknown size
* 574939 : [typefinding] flac typefinder mis-typefinds PDFs as flac ...
* 575550 : srt subtitle file keeps playbin2 from playing
* 575638 : kissfft copyright
* 575649 : [oggdemux] duration query in time format returns true wit...
* 576019 : On Windows queue2 can't write files longer than 2-4 GiB, ...
* 576142 : [vorbisenc] Non-header output buffers have NULL caps
* 576180 : [playbin2] Uses unref'd audiosink volume if using gconfau...
* 576586 : [alsamixer] gnome-sound-properties freeze
* 577054 : [videoscale] Not valgrind clean
* 577709 : Review new navigation API
* 577827 : [appsink] Have appsink new_buffer-callback return GstFlow...
* 578583 : [PATCH] multifdsink doesn't handle sync-method=latest-key...
* 578656 : Implement upstream GstForceKeyUnit events in theoraenc
* 579129 : pkgconfig: appsrc/appsink can not be linked to uninstalled
* 579130 : app: expose trivial type macros
* 579192 : gst_rtcp_packet_get_type should not assert on packet content
* 579203 : baseaudiosink: unparenting the ringbuffer in NULL causes ...
* 579267 : [rtspconnection] g_async_queue_new_full() is GLib-2.16 AP...
* 579463 : [cddabasesrc] [cdparanoiasrc] no longer emits discid
* 579668 : audioresample fails to build with --disable-gst-debug
* 579734 : [playbin] raw_decoding_mode seems to be set unconditionally
* 579912 : [decodebin2] multiqueue is too small in time (interleave ...
* 580470 : [audioresample] causes pipelines to go out of sync and be...
* 580952 : [audioresample] bad quality/pops compared to plughw
* 581727 : [playbin2] make playsink go to PAUSED async
* 569682 : playbin2 leaks request pad from input selector
* 580020 : [vorbisenc] causes buffers to be out of segment if new se...
* 562794 : rtspsrc fails to create a socket on Win32 sometimes.
* 567396 : playbin2: DECODE_BIN_LOCK occasionally called twice withi...
* 567982 : " queued_bytes " field isn't updated while flushing the que...
* 571299 : [appsink] Handoff callback API
* 574443 : rtsp win32 - forgotten variable
* 574516 : [typefind] add typefinder for photoshop .psd files
* 574964 : gst_app_src_end_of_stream(), mutex on error return
* 575256 : rtspsrc fails to resolve hostnames
* 575588 : decodebin2 deadlock
* 576187 : [playbin2] Stalls video sink when disabling subtitles in ...
* 576188 : [playbin2] Reusing a playbin2 instance with visualization...
* 576190 : [playbin2] Deadlock when reusing playbin2 after an error
* 577288 : " Internal playbin error " when seeking to the end of files
* 577610 : RTCP feedback messages support in GstRTCPPacket
* 577794 : [playbin2] leaks elements set through properties
* 578118 : [multifdsink] add option to not resend the streamheader w...
* 578506 : Pipeline with alsasrc and alsasink cannot change state ba...
* 578942 : Missing RTSP headers related to Windows Media extension.
* 580271 : videorate: fails to clear discont flag on duplicated buffers
* 580649 : uridecodebin: bug on documentation published in website
API added since 0.10.22:
* GstRTSP::gst_rtsp_options_as_text()
* GstRTSPMessage::gst_rtsp_message_take_header()
* GstRTSPRange::gst_rtsp_range_to_string()
* New Navigation interface commands, queries and messages
* gst_rtsp_channel_new()
* gst_rtsp_channel_unref()
* gst_rtsp_channel_attach()
* gst_rtsp_channel_queue_message()
* gst_rtsp_connection_accept()
* GstAppSink::gst_app_sink_set_callbacks()
* GST_VIDEO_FORMAT_YVYU,GST_VIDEO_BUFFER_TFF,GST_VIDEO_BUFFER_RFF,GST_VIDEO_BUFFER_ONEFIELD
* GST_MIXER_FLAG_HAS_WHITELIST,GST_MIXER_FLAG_GROUPING,GST_MIXER_TRACK_NO_RECORD,GST_MIXER_TRACK_NO_MUTE,GST_MIXER_TRACK_WHITELIST
* GstAppSrc::emit-signals
* GstAppSrc::gst_app_src_set_emit_signals()
* GstAppSrc::gst_app_src_get_emit_signals()
* GstAppSrc::gst_app_src_set_callbacks()
* RTSP::gst_rtsp_connection_get_url()
* GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
* RTSP:gst_rtsp_connection_set_tunneled()
* RTSP:gst_rtsp_connection_is_tunneled()
* RTSP::gst_rtsp_connection_set_ip()
* RTSP::gst_rtsp_connection_get_tunnelid()
* RTSP::gst_rtsp_connection_do_tunnel()
* RTSP::gst_rtsp_watch_reset()
IMPORTANT NOTES IMPORTANT NOTES
......
Release notes for GStreamer Base Plug-ins 0.10.22 "hidey hidey hidey ho" Release notes for GStreamer Base Plug-ins 0.10.23 "Emergency de-stress call"
...@@ -54,93 +54,127 @@ contains a set of less supported plug-ins that haven't passed the ...@@ -54,93 +54,127 @@ contains a set of less supported plug-ins that haven't passed the
Features of this release Features of this release
* Require gettext 0.17 * New navigation API to support DVD playback
* Replace audioresample with speexresample from -bad * playbin2 improvements
* Support new formats in RIFF: uncompressed RGB, WMA lossless, VP6 * RTSP extensions to allow extra headers and options
* Move libgstapp and elements from -bad * Replace audioresampler with speexresample based code
* Support color-key setting and probing for Xv properties * Support interlacing flags in the gstvideo library
* Improve typefinding for various formats * Support new RIFF formats
* Extend audio sinks for pull-mode operation * Improve typefinding
* Support for more subtitle formats * Support more frame formats in videoscale
* More development on decode2bin and playbin2 * Many other bug-fixes and improvements
* RTP and SDP fixes
* Many bug fixes and improvements
Bugs fixed in this release Bugs fixed in this release
* 562163 : theoraenc likely ignoring segments * 577637 : [playbin2] expose temp-location property
* 562258 : rtspsrc element takes long time to error out if the addre... * 580120 : [playbin2] unit test fails
* 561789 : [volume] deadlocks with a controller attached * 478512 : [alsamixer] volume control slider not working
* 554533 : [xvimagesink] allow setting colorkey if possible * 574962 : rhythmbox crash in flac_type_find
* 567511 : colorkey in xvimagesink gets reset when element is reused
* 116051 : libresample doesn't handle > factor of 2 rate conversion
* 346218 : [audioresample] doesn't do anti aliasing
* 385061 : [audioresample?] investigate high CPU usage
* 456788 : [subparse] can't handle UTF-16 charset encoded subtitle.
* 525807 : [vorbisenc] vorbisenc has problems with a gnlsource that ...
* 546955 : gstoggmux EOS handling issue
* 549417 : [audioresample] unit test fails on 64bit linux
* 549510 : audioresample doesn't negotiate ideal caps
* 552237 : UTF-16 srt confuses gstreamer, misdetected as mp3
* 552559 : Implementation of SLAVE_SKEW in baseaudiosrc
* 552569 : audioresample producing strange sized buffers
* 552801 : audioconvert can overflow with big audio buffers
* 554879 : Add ability to specify format for date/time display in Gs...
* 555257 : Doesn't display srt subtitles saved with BOM
* 555319 : add FFV1 fourcc to riff-media
* 555607 : subrip subtitles typefind too strict
* 555699 : [PATCH] theoradec: prefer container's pixel aspect ratio ...
* 556025 : build failure in tests/icles
* 556066 : Last byte of FLAC image buffer chopped off
* 557365 : subparse check fails
* 558124 : [PLUGIN-MOVE] Move speexresample as audioresample2 to -base
* 559111 : ALSA sink hangs on USB audio device unplug while playing
* 559478 : does not play windows media streams correctly
* 559567 : `gst_base_audio_sink_sync_latency' should call `gst_base_...
* 561436 : videorate element add image/jpeg to caps template
* 561734 : playbin2 additions
* 561780 : Playbin2 should work without volume too
* 561924 : oggdemux hangs when given corrupt input via non-seekable ...
* 562270 : build without gdk fails
* 563143 : ximagesink/xvimagesink : _alloc_buffer returns non-clean ...
* 563174 : Implement gst_rtcp_packet_remove
* 563508 : [rgvolume] Unit test fails with passthrough assertions
* 563718 : Theora check out of date
* 563904 : GNOME Goal: Clean up GLib and GTK+ includes
* 564098 : MS Word files are recognised as audio/mpeg and OSX's .DS_...
* 564139 : Documentation of TCP plugins * 564139 : Documentation of TCP plugins
* 564200 : GstBaseAudioSink should register its enums and have corre... * 577436 : xvimagesink should use xcontext- > depth and not count bits...
* 564206 : GstBaseAudioSrc should register its enum and have corresp... * 350311 : [playbin2] support for subpicture subtitles
* 564421 : Move appsrc/appsink to -base * 378094 : Enable pango elements to handle UYVY
* 564929 : Audiosink blocks if setcaps called while playing * 543591 : Gnonlin can not play theora streams
* 566586 : playbin2 test7.c fails after two songs * 553295 : [riff] fuzzed AVI file causes segfault
* 566750 : [appsrc/sink] add padding, move private data to private s... * 565105 : Gstreamer does not change from READY back to PAUSED in sa...
* 566761 : [gstapp] No pkg-config file * 565777 : [riff] unrecognised video fourcc 0x10000002 for mpeg2 in avi
* 566837 : gst_cdda_base_src_mode_get_type() is not public from < gst... * 566661 : [typefind] Fall back to file extension using uri query
* 566875 : [gnomevfs] Add dependency for the GnomeVFS modules * 567255 : [riff] doesn't detect codec_id 0x706d as AAC (amongst other)
* 566876 : [gio] Add dependency for the modules dir * 567636 : [pbutils] Missing plugins code shouldn't ask for the same...
* 567027 : Add GType for GstRTSPUrl for bindings * 567740 : bogus warning in decodebin2?
* 567168 : appsink is using the wrong signal slot for the pull-buffe... * 568482 : linking problems in gst-plugins-base
* 567960 : [tagdemux] Doesn't forward unknown events upstream * 569655 : [ffmpegcolorspace] Add UYVY422 to GRAY8 conversion function
* 500833 : [FFT] Struct alignment issues on sparc * 570142 : Documentation is broken for uridecodebin
* 552199 : Parsing SDP file with multicast address fails * 570356 : aac typefinder failure
* 558553 : [riff] gst_riff_create_video_caps not recognizing certain... * 570768 : [ximagesink] wrong mouse pointer position if output windo...
* 564896 : gst_netaddress_get_ip[46]_address should check for correc... * 570832 : Add flags to enhance mixer interfaces
* 566341 : Some Ogg Theora files don't finished at seek at the end * 571009 : [tagdemux] WMA file with id3v2 tag causes assertion to fail
* 566654 : playbin2: does not come back from NULL after switching UR... * 571147 : [ffmpegcolorspace/videotestsrc] Add support for packed/pl...
* 566723 : GstAudioClock's new function may better use const gchar* ... * 572577 : [playbin2] deadlock on shutdown
* 572872 : [ffmpegcolorspace] Add YVYU colorspace
* 572993 : [subparse] broken libregex dependency on Windows
* 573165 : Generate additional export files for gstreamer app plugin
* 573528 : Wrong format modifier in gstgiobasesink.c
* 573529 : In gstrtspconnection.c some functions are called with wro...
* 574293 : [decodebin2] deadlock on shutdown
* 574319 : Missing HAVE_PROCESS_H in win32/common/config.h
* 574447 : gstadder.c: line 904: error C2036: 'gpointer' : unknown size
* 574939 : [typefinding] flac typefinder mis-typefinds PDFs as flac ...
* 575550 : srt subtitle file keeps playbin2 from playing
* 575638 : kissfft copyright
* 575649 : [oggdemux] duration query in time format returns true wit...
* 576019 : On Windows queue2 can't write files longer than 2-4 GiB, ...
* 576142 : [vorbisenc] Non-header output buffers have NULL caps
* 576180 : [playbin2] Uses unref'd audiosink volume if using gconfau...
* 576586 : [alsamixer] gnome-sound-properties freeze
* 577054 : [videoscale] Not valgrind clean
* 577709 : Review new navigation API
* 577827 : [appsink] Have appsink new_buffer-callback return GstFlow...
* 578583 : [PATCH] multifdsink doesn't handle sync-method=latest-key...
* 578656 : Implement upstream GstForceKeyUnit events in theoraenc
* 579129 : pkgconfig: appsrc/appsink can not be linked to uninstalled
* 579130 : app: expose trivial type macros
* 579192 : gst_rtcp_packet_get_type should not assert on packet content
* 579203 : baseaudiosink: unparenting the ringbuffer in NULL causes ...
* 579267 : [rtspconnection] g_async_queue_new_full() is GLib-2.16 AP...
* 579463 : [cddabasesrc] [cdparanoiasrc] no longer emits discid
* 579668 : audioresample fails to build with --disable-gst-debug
* 579734 : [playbin] raw_decoding_mode seems to be set unconditionally
* 579912 : [decodebin2] multiqueue is too small in time (interleave ...
* 580470 : [audioresample] causes pipelines to go out of sync and be...
* 580952 : [audioresample] bad quality/pops compared to plughw
* 581727 : [playbin2] make playsink go to PAUSED async
* 569682 : playbin2 leaks request pad from input selector
* 580020 : [vorbisenc] causes buffers to be out of segment if new se...
* 562794 : rtspsrc fails to create a socket on Win32 sometimes.
* 567396 : playbin2: DECODE_BIN_LOCK occasionally called twice withi...
* 567982 : " queued_bytes " field isn't updated while flushing the que...
* 571299 : [appsink] Handoff callback API
* 574443 : rtsp win32 - forgotten variable
* 574516 : [typefind] add typefinder for photoshop .psd files
* 574964 : gst_app_src_end_of_stream(), mutex on error return
* 575256 : rtspsrc fails to resolve hostnames
* 575588 : decodebin2 deadlock
* 576187 : [playbin2] Stalls video sink when disabling subtitles in ...
* 576188 : [playbin2] Reusing a playbin2 instance with visualization...
* 576190 : [playbin2] Deadlock when reusing playbin2 after an error
* 577288 : " Internal playbin error " when seeking to the end of files
* 577610 : RTCP feedback messages support in GstRTCPPacket
* 577794 : [playbin2] leaks elements set through properties
* 578118 : [multifdsink] add option to not resend the streamheader w...
* 578506 : Pipeline with alsasrc and alsasink cannot change state ba...
* 578942 : Missing RTSP headers related to Windows Media extension.
* 580271 : videorate: fails to clear discont flag on duplicated buffers
* 580649 : uridecodebin: bug on documentation published in website
API changed in this release API changed in this release
- API additions: - API additions:
* clockoverlay::time-format * GstRTSP::gst_rtsp_options_as_text()
* GstRingBuffer:gst_ring_buffer_activate() * GstRTSPMessage::gst_rtsp_message_take_header()
* GstRingBuffer:gst_ring_buffer_is_active() * GstRTSPRange::gst_rtsp_range_to_string()
* GstRingBuffer:gst_ring_buffer_convert() * New Navigation interface commands, queries and messages
* Add GST_TYPE_BASE_AUDIO_(SRC|SINK)_SLAVE_METHOD to the public API * gst_rtsp_channel_new()
* gst_netaddress_get_address_bytes() * gst_rtsp_channel_unref()
* gst_netaddress_set_address_bytes() * gst_rtsp_channel_attach()
* gst_rtsp_channel_queue_message()
* gst_rtsp_connection_accept()
* GstAppSink::gst_app_sink_set_callbacks()
* GST_VIDEO_FORMAT_YVYU,GST_VIDEO_BUFFER_TFF,GST_VIDEO_BUFFER_RFF,GST_VIDEO_BUFFER_ONEFIELD
* GST_MIXER_FLAG_HAS_WHITELIST,GST_MIXER_FLAG_GROUPING,GST_MIXER_TRACK_NO_RECORD,GST_MIXER_TRACK_NO_MUTE,GST_MIXER_TRACK_WHITELIST
* GstAppSrc::emit-signals
* GstAppSrc::gst_app_src_set_emit_signals()
* GstAppSrc::gst_app_src_get_emit_signals()
* GstAppSrc::gst_app_src_set_callbacks()
* RTSP::gst_rtsp_connection_get_url()
* GstRTSPLowerTrans::GST_RTSP_LOWER_TRANS_HTTP
* RTSP:gst_rtsp_connection_set_tunneled()
* RTSP:gst_rtsp_connection_is_tunneled()
* RTSP::gst_rtsp_connection_set_ip()
* RTSP::gst_rtsp_connection_get_tunnelid()
* RTSP::gst_rtsp_connection_do_tunnel()
* RTSP::gst_rtsp_watch_reset()
Download Download
...@@ -159,8 +193,7 @@ http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer ...@@ -159,8 +193,7 @@ http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer
Developers Developers
CVS is hosted on cvs.freedesktop.org. GStreamer is stored in Git, hosted at git.freedesktop.org, and can be cloned from there.
All code is in CVS and can be checked out from there.
Interested developers of the core library, plug-ins, and applications should Interested developers of the core library, plug-ins, and applications should
subscribe to the gstreamer-devel list. If there is sufficient interest we subscribe to the gstreamer-devel list. If there is sufficient interest we
will create more lists as necessary. will create more lists as necessary.
...@@ -170,38 +203,37 @@ Applications ...@@ -170,38 +203,37 @@ Applications
Contributors to this release Contributors to this release
* Alessandro Decina
* Andrew Feren
* Andy Wingo * Andy Wingo
* Antoine Tremblay
* Benjamin Gaignard
* Benjamin M. Schwartz
* Brian Cameron
* Christian Schaller * Christian Schaller
* Cygwin Ports maintainer * David Flynn
* Damien Lespiau
* Daniel Drake
* David Schleef * David Schleef
* Edward Hervey * Edward Hervey
* Guillaume Emont * Felipe Contreras
* Håvard Graff * Garret D'Amore
* Jan Gerber * Hannes Bistry
* Jan Schmidt * Jan Schmidt
* Jan Urbanski
* Johann Prieur
* Jonas Danielsson
* Jonathan Matthew * Jonathan Matthew
* Jonathan Rosser * Josep Torra
* José Alburquerque
* Julien Moutte * Julien Moutte
* Klaas * Luca Ognibene
* Luis Menina
* Mark Nauwelaerts * Mark Nauwelaerts
* Matthias Kretz * Martin Samuelsson
* Michael Smith * Michael Smith
* Nick Haddad
* Olivier Crete * Olivier Crete
* Pavel Zeldin * Peter Kjellerstedt
* Robin Stocker * René Stadler
* Sebastian Dröge * Sebastian Dröge
* Stefan Kost * Stefan Kost
* Tero Saarni
* Thomas Vander Stichele
* Tim-Philipp Müller * Tim-Philipp Müller
* Tomas Hoger
* Wim Taymans * Wim Taymans
* xavierb at gmail dot com * Zaheer Merali
* 이문형 * Zeeshan Ali
   
\ No newline at end of file
...@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file ...@@ -5,7 +5,7 @@ dnl please read gstreamer/docs/random/autotools before changing this file
dnl initialize autoconf dnl initialize autoconf
dnl releases only do -Wall, cvs and prerelease does -Werror too dnl releases only do -Wall, cvs and prerelease does -Werror too
dnl use a three digit version number for releases, and four for cvs/prerelease dnl use a three digit version number for releases, and four for cvs/prerelease
AC_INIT(GStreamer Base Plug-ins, 0.10.22.6, AC_INIT(GStreamer Base Plug-ins, 0.10.23,
http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer, http://bugzilla.gnome.org/enter_bug.cgi?product=GStreamer,
gst-plugins-base) gst-plugins-base)
......
...@@ -838,6 +838,26 @@ ...@@ -838,6 +838,26 @@
<DEFAULT>-1</DEFAULT> <DEFAULT>-1</DEFAULT>
</ARG> </ARG>
<ARG>
<NAME>GstMultiFdSink::handle-read</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Handle Read</NICK>
<BLURB>Handle client reads and discard the data.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG>
<NAME>GstMultiFdSink::resend-streamheader</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Resend streamheader</NICK>
<BLURB>Resend the streamheader if it changes in the caps.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG> <ARG>
<NAME>GstDecodeBin::threaded</NAME> <NAME>GstDecodeBin::threaded</NAME>
<TYPE>gboolean</TYPE> <TYPE>gboolean</TYPE>
...@@ -2378,6 +2398,16 @@ ...@@ -2378,6 +2398,16 @@
<DEFAULT>-1</DEFAULT> <DEFAULT>-1</DEFAULT>
</ARG> </ARG>
<ARG>
<NAME>GstPlayBin2::text-sink</NAME>
<TYPE>GstElement*</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Text plugin</NICK>
<BLURB>the text output element to use (NULL = default textoverlay).</BLURB>
<DEFAULT></DEFAULT>
</ARG>
<ARG> <ARG>
<NAME>GstGioSink::location</NAME> <NAME>GstGioSink::location</NAME>
<TYPE>gchar*</TYPE> <TYPE>gchar*</TYPE>
...@@ -2748,6 +2778,16 @@ ...@@ -2748,6 +2778,16 @@
<DEFAULT>Stream</DEFAULT> <DEFAULT>Stream</DEFAULT>
</ARG> </ARG>
<ARG>
<NAME>GstAppSrc::emit-signals</NAME>
<TYPE>gboolean</TYPE>
<RANGE></RANGE>
<FLAGS>rw</FLAGS>
<NICK>Emit signals</NICK>
<BLURB>Emit new-preroll and new-buffer signals.</BLURB>
<DEFAULT>TRUE</DEFAULT>
</ARG>
<ARG> <ARG>
<NAME>GstAudioResample::filter-length</NAME> <NAME>GstAudioResample::filter-length</NAME>
<TYPE>gint</TYPE> <TYPE>gint</TYPE>
......
...@@ -15,17 +15,43 @@ GObject ...@@ -15,17 +15,43 @@ GObject
GstDecodeBin2 GstDecodeBin2
GstURIDecodeBin GstURIDecodeBin
GstDecodeBin GstDecodeBin
GstBaseSink
GstGioBaseSink
GstGioSink
GstGioStreamSink
GstGnomeVFSSink
GstBaseAudioSink
GstAudioSink
GstAlsaSink
GstVideoSink
GstXvImageSink
GstXImageSink
GstTCPClientSink
GstMultiFdSink
GstTCPServerSink
GstAppSink
GstBaseSrc
GstGioBaseSrc
GstGioSrc
GstGioStreamSrc
GstPushSrc
GstCddaBaseSrc
GstCdParanoiaSrc
GstBaseAudioSrc
GstAudioSrc
GstAlsaSrc
GstV4lElement
GstV4lSrc
GstTCPClientSrc
GstTCPServerSrc
GstVideoTestSrc
GstGnomeVFSSrc
GstAppSrc
GstAudioTestSrc
GstVorbisEnc GstVorbisEnc
GstVorbisDec GstVorbisDec
GstVorbisParse GstVorbisParse
GstVorbisTag GstVorbisTag
GstTheoraDec
GstTheoraEnc
GstTheoraParse
GstTextOverlay
GstTimeOverlay
GstClockOverlay
GstTextRender
GstOggDemux GstOggDemux
GstOggMux GstOggMux
GstOgmParse GstOgmParse
...@@ -35,64 +61,38 @@ GObject ...@@ -35,64 +61,38 @@ GObject
GstOggParse GstOggParse