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Release notes for GStreamer Base Plug-ins 0.10.23 "Emergency de-stress call"

The GStreamer team is proud to announce a new release in the 0.10.x stable series of the GStreamer Base Plug-ins.

The 0.10.x series is a stable series targeted at end users. It is not API or ABI compatible with the stable 0.8.x series. It is, however, parallel installable with the 0.8.x series.

This module contains a set of reference plugins, base classes for other plugins, and helper libraries. This module is kept up-to-date together with the core developments. Element writers should look at the elements in this module as a reference for their development. This module contains elements for, among others:

  • device plugins: x(v)imagesink, alsa, v4lsrc, cdparanoia
  • containers: ogg
  • codecs: vorbis, theora
  • text: textoverlay, subparse
  • sources: audiotestsrc, videotestsrc, gnomevfssrc
  • network: tcp
  • typefind
  • audio processing: audioconvert, adder, audiorate, audioscale, volume
  • visualisation: libvisual
  • video processing: ffmpegcolorspace
  • aggregate elements: decodebin, playbin
Other modules containing plug-ins are:
gst-plugins-good
contains a set of well-supported plug-ins under our preferred license
gst-plugins-ugly
contains a set of well-supported plug-ins, but might pose problems for distributors
gst-plugins-bad
contains a set of less supported plug-ins that haven't passed the rigorous quality testing we expect

Features of this release

  • 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 in this release

  • 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 changes

  • API additions
    • 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()

Download

You can find source releases of gst-plugins-base in the gst-plugins-base download directory.

The git repository and details how to clone it can be found at git.freedesktop.org.

Homepage

The project's website is http://gstreamer.freedesktop.org.

Support and Bugs

We use GNOME's bugzilla for bug reports and feature requests.

Please submit patches via bugzilla as well.

For help and support, please subscribe to and send questions to the gstreamer-devel mailing list (see below for details).

There is also a #gstreamer IRC channel on the Freenode IRC network.

Developers

Git is hosted on git.freedesktop.org. You can browse the gst-plugins-base repository.

All code is in Git and can be checked out from there.

Interested developers of the core library, plugins, and applications should subscribe to the gstreamer-devel list.

Applications

Contributors to this release

  • Andy Wingo
  • Antoine Tremblay
  • Benjamin Gaignard
  • Benjamin M. Schwartz
  • Brian Cameron
  • Christian Schaller
  • David Flynn
  • David Schleef
  • Edward Hervey
  • Felipe Contreras
  • Garret D'Amore
  • Hannes Bistry
  • Jan Schmidt
  • Jan Urbanski
  • Johann Prieur
  • Jonas Danielsson
  • Jonathan Matthew
  • Josep Torra
  • Julien Moutte
  • Luca Ognibene
  • Mark Nauwelaerts
  • Martin Samuelsson
  • Michael Smith
  • Olivier Crete
  • Peter Kjellerstedt
  • René Stadler
  • Sebastian Dröge
  • Stefan Kost
  • Tim-Philipp Müller
  • Tomas Hoger
  • Wim Taymans
  • Zaheer Merali
  • Zeeshan Ali

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