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open source multimedia framework
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Release notes for GStreamer Base Plugins 1.3.1

The GStreamer team is pleased to announce the first release of the unstable 1.3 release series. The 1.3 release series is adding new features on top of the 1.0 and 1.2 series and is part of the API and ABI-stable 1.x release series of the GStreamer multimedia framework. The unstable 1.3 release series will lead to the stable 1.4 release series in the next weeks, and newly added API can still change until that point.

Binaries for Android, iOS, Mac OS X and Windows will be provided separately during the unstable 1.3 release series.

The versioning scheme that is used in general is that 1.x.y is API and ABI backwards compatible with previous 1.x.y releases. If x is an even number it is a stable release series and all releases in this series will only contain important bugfixes, e.g. the 1.0 series with 1.0.7. If x is odd it is a development release series that will lead to the next stable release series 1.x+1 and contains new features and bigger changes. During the development release series, new API can still change.

This module contains a set of reference plugins, base classes for other plugins, and helper libraries. It also includes essential elements such as audio and video format converters, and higher-level components like playbin, decodebin, encodebin, and discoverer. 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, giosrc
  • network: tcp
  • typefind functions
  • audio processing: audioconvert, adder, audiorate, audioresample, volume
  • visualisation: libvisual
  • video processing: videoconvert, videoscale
  • high-level components: playbin, uridecodebin, decodebin, encodebin, discoverer
  • libraries: app, audio, fft, pbutils, riff, rtp, rtsp, sdp, tag, video
Other modules containing plugins are:
gst-plugins-good
contains a set of well-supported plugins under our preferred license
gst-plugins-ugly
contains a set of well-supported plugins, but might pose problems for distributors
gst-plugins-bad
contains a set of less supported plugins that haven't passed the rigorous quality testing we expect, or are still missing documentation and/or unit tests
gst-libav
contains a set of codecs plugins based on libav (formerly gst-ffmpeg)

Bugs fixed in this release

  • 684030 : typefinding: mp4 with video and dts ES detected as DTS audio
  • 725078 : audiobasesink: clip start samples to match clipped timestamp from skew algorithm
  • 708633 : adder: Should not take channel mask in consideration when in mono or stereo
  • 540941 : v4l2: RGB32 should be mapped to xRGB instead of RGBx
  • 646577 : rtppayload: Make RTP time information accessible
  • 670690 : audioresample: missing configure checks for SSE / SSE2
  • 678402 : Device discovery/listing replacement for GstPropertyProbe
  • 678590 : subparse: Add support for LRC subtitles
  • 679031 : playbin/playsink: Add support for audio and video filters
  • 687183 : videodecoder: Allow to negotiate a buffer pool before output format is known
  • 702230 : audioringbuffer: Don't access timestamps array if not acquired
  • 707361 : video: Add support for 64x32 tiled NV12 color format
  • 707636 : dashdemux: offline playback not buffering correctly
  • 708680 : typefind: Add typefind function for H265
  • 708921 : pbutils: Add codec-utility functions to support h265
  • 708991 : audiocdsrc: invalid musicbrainz discids because of trailing data tracks
  • 709588 : encodebin: Handle changes in encoding_profile::restriction during playback
  • 709646 : videotestsrc: Could implement duration query when num-buffers is set
  • 709755 : alsa: add channel map API support
  • 709814 : [examples/overlay] avoid to unref sink if not found. Also fix logic to find a sink in one of the example.
  • 709858 : theoraenc: Do nothing when flushing the encoder when no caps were set
  • 710760 : videoconvert: remove unneeded guint comparison
  • 711094 : videodecoder: improve max-error handling
  • 711258 : sdp: fix duplicate 'const' declaration warnings
  • 712798 : videometa: add GstVideoGLTextureUploadMeta buffer pool option
  • 719383 : rtpbasepayload: Perfect timestamps confusingly explained
  • 719415 : rtpbasepayload: Expose running time of last processed buffer
  • 719850 : convertframe: remove trivial memory leak
  • 719890 : videodecoder: Add API to get the currently pending, parsed frame size
  • 720103 : videodecoder: Introduce sink_query/src_query
  • 720124 : tests/examples/overlay/qt-videooverlay.cpp has incorrect include from Qt
  • 720162 : tests: Add test for rtpbasepayload/-depayload
  • 720205 : playback: add video/x-raw(ANY) to default raw caps
  • 720215 : sdp: parse encryption key field
  • 720219 : rtsptransport: allow getting mime type by profile
  • 720389 : videodecoder: should release buffer pool sooner
  • 720810 : audio/video: Initialize all {audio|video}info fields
  • 720999 : Missing annotation for GstColorBalance interface
  • 721103 : test-effect-switch errors out with not-negotiated after a while
  • 721701 : videoconvert: I420 to BGRA conversion is slower than in 0.10
  • 721953 : pango: basetextoverlay: handle video/x-raw(ANY) if downstream supports the GstVideoOverlayCompositionMeta API
  • 722330 : streamsplitter: negotiation problems with parsers
  • 722491 : playbin: remove duplicate assignment
  • 722682 : oggmux: problems with vp8 stream
  • 723096 : decodebin: Make it possible to register multiple handlers to decodebin's autoplug-select signal
  • 723271 : videotestsrc: fix a warning if downstream does not propose a buffer pool
  • 723328 : gstrtpbase(|de)payload: add more unit tests and fix bugs
  • 723492 : gst-plugins-base: Do not build check tests for disabled plugins
  • 723507 : jsseek: Add missing HAVE_X check
  • 724393 : rtspconnection: allow specifying an anchor certificate database
  • 724509 : audioconvert: outputs silence when converting certain mono caps to certain other mono caps
  • 724828 : playbin: improve autoplug_query_caps return
  • 724893 : playsinkconvertbin: improve gst_play_sink_convert_bin_getcaps return
  • 725034 : all plugin sets but -base don't install gtk-doc docs without '--enable-gtk-doc'
  • 725206 : rtspconnection: Missing include file
  • 725479 : gst-plugins-base: Ignore gcov intermediate files
  • 725521 : docs: Fix argument and annotation typos, add missing annotations and remove duplicate section
  • 725658 : Removing some GnomeVFS left bits
  • 725837 : pango: textoverlay: lot of warnings in debug log with framerate=0/1
  • 725878 : rtspconnection: headers in GET response not configurable for tunnels
  • 725898 : Lose data when producing data faster than sendt during tunneling rtps/rtp(TCP)
  • 726433 : rtspconnection: setsockopt() argument 4 is not properly casted for W32
  • 726641 : rtspconnection: connection_poll() not working correctly
  • 727498 : videodecoder: deactivates downstream bufferpool
  • 728772 : rtspconnection: stuck in teardown
  • 728845 : gst-play: add option to supply input media-files from a playlist file
  • 728907 : rtspconnection: add more tests
  • 729114 : audiodecoder: default caps nego will manually fixate non-mutable caps
  • 729117 : rtpbuffer: fix memory leak when gst_rtp_buffer_map fails
  • 729195 : videotestsrc: undefined behaviour in left-shift
  • 729321 : playbin/subtitleoverlay: Deadlock when changing subtitle track while PAUSED
  • 704933 : uridecodebin: allow progressive buffering with more media types

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 https://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.

Contributors to this release

  • Adrien Schwartzentruber
  • Aleix Conchillo Flaque
  • Aleix Conchillo Flaqué
  • Alessandro Decina
  • Andres Gomez
  • Antoine Jacoutot
  • Antonio Ospite
  • Arun Raghavan
  • Bastien Nocera
  • Christian Fredrik Kalager Schaller
  • David Svensson Fors
  • Edward Hervey
  • Eric Trousset
  • George Kiagiadakis
  • Göran Jönsson
  • Haakon Sporsheim
  • Hans Månsson
  • Holger Kaelberer
  • Jan Schmidt
  • Jihyun Cho
  • Johannes Dewender
  • John Bassett
  • Josep Torra
  • Julien Isorce
  • Justin Joy
  • Lionel Landwerlin
  • Luis de Bethencourt
  • Mark Nauwelaerts
  • Matej Knopp
  • Mathieu Duponchelle
  • MathieuDuponchelle
  • Matthew Waters
  • Matthieu Bouron
  • Nicola Murino
  • Nicolas Dufresne
  • Ognyan Tonchev
  • Olivier Crête
  • Rafał Mużyło
  • Ravi Kiran K N
  • Reynaldo H. Verdejo Pinochet
  • Sebastian Dröge
  • Sebastian Rasmussen
  • Sjoerd Simons
  • Sreerenj Balachandran
  • Stefan Sauer
  • Stephan Sundermann
  • Stian Selnes
  • Stéphane Cerveau
  • Takashi Iwai
  • Thiago Santos
  • Thibault Saunier
  • Tim-Philipp Müller
  • Todd Agulnick
  • Tom Greenwood
  • Vincent Penquerc'h
  • William Grant
  • Wim Taymans
  • Wonchul Lee
  • Руслан Ижбулатов

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