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Release notes for GStreamer Base Plugins 1.5.90

The GStreamer team is pleased to announce the first release candidate for the stable 1.6 release series. The 1.6 release series is adding new features on top of the 1.0, 1.2 and 1.4 series and is part of the API and ABI-stable 1.x release series of the GStreamer multimedia framework. The final 1.6.0 release is planned in the next few days unless any major bugs are found.

Binaries for Android, iOS, Mac OS X and Windows will be provided separately by the GStreamer project.

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

  • 728636 : textoverlay: cuts off right edge italicised text
  • 737427 : appsink: Can't influence allocation query to satisfy user needs
  • 742385 : video/audio encoders/decoders: need API to determine when to copy over GstMetas and when to drop them
  • 746010 : oggdemux: doesn't go into pull mode even when queue2 ring-buffer is enabled
  • 748635 : videorate: caps negotiation regression
  • 750235 : [API] rtpbasedepayload: add process_rtp_packet() vfunc
  • 751147 : appsink: pull_preroll returns wrong segment in the sample
  • 751690 : playbin : remove unnecessary break
  • 751883 : rtcpbuffer: Fix validation of packets with padding
  • 751932 : GstVideoDecoder: Fix setting default pixel-aspect-ratio
  • 752011 : video: Add boxed type for GstVideoInfo structure
  • 752040 : gst-play: fix memory leak
  • 752041 : ximagesink: fix navigation event leak
  • 752042 : xvimagesink: fix pad memory leak
  • 752043 : ximagesink: Post navigation events as message on the bus
  • 752051 : GstVideoScaler: Initialised scaling functions to get rid of compiler messages
  • 752059 : xvimagesink: refactor to use gst_pad_push_event
  • 752111 : rtpbasedepayload: Fix minor leak of segment event
  • 752156 : playsink: Require streamvolume interface for sink volumes for standardized behaviour
  • 752228 : payloader/depayload: Wrong segment handling
  • 752359 : playback-test: fix memory leak
  • 752397 : playback-test: fix tag list leak
  • 752400 : jsseek: fix tag list leak
  • 752436 : typefind: Treat *.umx (Unreal Music Package) as audio/x-mod
  • 752454 : jsseek: fix memory leaks
  • 753001 : rtp buffer: NULL GstBuffer pointer being passed within gst_rtp_buffer_set_extension_data
  • 753078 : oggdemux: Segfault on state-change intensive test
  • 753088 : decodebin: fix deadend_details string leak
  • 753360 : videodecoder: pushes gap before segment
  • 753483 : videorate: add support for bayer formats
  • 753701 : discoverer: Few trivial fixes in handling error cases
  • 611157 : video: API to signal stereoscopic and multiview video

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).

Find us on IRC at #gstreamer.

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

  • Brian Peters
  • Edward Hervey
  • George Kiagiadakis
  • Hyunjun Ko
  • Jan Schmidt
  • Joan Pau Beltran
  • Lubosz Sarnecki
  • Luis de Bethencourt
  • Marcin Kolny
  • Myoungsun Lee
  • Nicolas Dufresne
  • Olivier Crête
  • Rico Tzschichholz
  • Sebastian Dröge
  • Stefan Sauer
  • Stian Selnes
  • Thiago Santos
  • Tim-Philipp Müller
  • Tobias Mueller
  • Ville Skyttä
  • Vineeth T M
  • Vineeth TM
  • Vivia Nikolaidou
  • Wim Taymans
  • danny song

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